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Versions: (RFC 2326) 00 01 02 03 04 05 06 07 08 09 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 34 35 36 37 38 39 40 RFC 7826

Internet Engineering Task Force                               MMUSIC WG
Internet Draft                                           H. Schulzrinne
draft-ietf-mmusic-rfc2326bis-07.txt                         Columbia U.
July 19, 2004                                                    A. Rao
Expires: January, 2005                                            Cisco
                                                            R. Lanphier
                                                           RealNetworks
                                                          M. Westerlund
                                                               Ericsson
                                                          A. Narasimhan
                                                              Princeton



                  Real Time Streaming Protocol (RTSP)


STATUS OF THIS MEMO

   By submitting this Internet-Draft, I (we) certify that any applicable
   patent or other IPR claims of which I am (we are) aware have been
   disclosed, and any of which I (we) become aware will be disclosed, in
   accordance with RFC 3668 (BCP 79).

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups.  Note that
   other groups may also distribute working documents as Internet-
   Drafts.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress".

   The list of current Internet-Drafts can be accessed at
   http://www.ietf.org/ietf/1id-abstracts.txt

   To view the list Internet-Draft Shadow Directories, see
   http://www.ietf.org/shadow.html.

Abstract

   This memorandum is a revision of RFC 2326, which is currently a
   Proposed Standard.

   The Real Time Streaming Protocol, or RTSP, is an application-level
   protocol for control over the delivery of data with real-time
   properties. RTSP provides an extensible framework to enable
   controlled, on-demand delivery of real-time data, such as audio and
   video. Sources of data can include both live data feeds and stored
   clips. This protocol is intended to control multiple data delivery



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   sessions, provide a means for choosing delivery channels such as UDP,
   multicast UDP and TCP, and provide a means for choosing delivery
   mechanisms based upon RTP (RFC 3550).
















































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                           Table of Contents



   1          Introduction ........................................    9
   1.1        RTSP Specification Update ...........................    9
   1.2        Purpose .............................................   10
   1.3        Notational Conventions ..............................   11
   1.4        Terminology .........................................   12
   1.5        Protocol Properties .................................   15
   1.6        Extending RTSP ......................................   17
   1.7        Overall Operation ...................................   18
   1.8        RTSP States .........................................   19
   1.9        Relationship with Other Protocols ...................   20
   2          RTSP Use Cases ......................................   20
   2.1        On-demand Playback of Stored Content ................   20
   2.2        Unicast distribution of Live Content ................   22
   2.3        On-demand Playback using Multicast ..................   22
   2.4        Inviting a RTSP server into a conference ............   23
   2.5        Live Content using Multicast ........................   24
   3          Protocol Parameters .................................   25
   3.1        RTSP Version ........................................   25
   3.2        RTSP URL ............................................   25
   3.3        Session Identifiers .................................   26
   3.4        SMPTE Relative Timestamps ...........................   27
   3.5        Normal Play Time ....................................   27
   3.6        Absolute Time .......................................   28
   3.7        Feature-tags ........................................   28
   3.8        Entity Tags .........................................   29
   4          RTSP Message ........................................   29
   4.1        Message Types .......................................   29
   4.2        Message Headers .....................................   30
   4.3        Message Body ........................................   30
   4.4        Message Length ......................................   30
   5          General Header Fields ...............................   30
   6          Request .............................................   30
   6.1        Request Line ........................................   31
   6.2        Request Header Fields ...............................   32
   7          Response ............................................   32
   7.1        Status-Line .........................................   33
   7.1.1      Status Code and Reason Phrase .......................   33
   7.1.2      Response Header Fields ..............................   35
   8          Entity ..............................................   35
   8.1        Entity Header Fields ................................   35
   8.2        Entity Body .........................................   35



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   9          Connections .........................................   36
   9.1        Pipelining ..........................................   36
   9.2        Reliability and Acknowledgements ....................   36
   9.3        Unreliable Transport ................................   39
   9.4        The usage of connections ............................   39
   9.5        Timing Out RTSP messages ............................   41
   9.6        Use of IPv6 .........................................   41
   10         Capability Handling .................................   42
   11         Method Definitions ..................................   43
   11.1       OPTIONS .............................................   44
   11.2       DESCRIBE ............................................   45
   11.3       SETUP ...............................................   47
   11.4       PLAY ................................................   50
   11.5       PAUSE ...............................................   54
   11.6       TEARDOWN ............................................   58
   11.7       GET_PARAMETER .......................................   58
   11.8       SET_PARAMETER .......................................   59
   11.9       REDIRECT ............................................   61
   11.10      PING ................................................   63
   12         Embedded (Interleaved) Binary Data ..................   64
   13         Status Code Definitions .............................   65
   13.1       Success 1xx .........................................   66
   13.1.1     100 Continue ........................................   66
   13.2       Success 2xx .........................................   66
   13.3       Redirection 3xx .....................................   66
   13.3.1     TBW .................................................   66
   13.3.2     301 Moved Permanently ...............................   66
   13.3.3     302 Found ...........................................   66
   13.3.4     303 See Other .......................................   67
   13.3.5     304 Not Modified ....................................   67
   13.3.6     305 Use Proxy .......................................   67
   13.4       Client Error 4xx ....................................   67
   13.4.1     400 Bad Request .....................................   67
   13.4.2     405 Method Not Allowed ..............................   68
   13.4.3     451 Parameter Not Understood ........................   68
   13.4.4     452 reserved ........................................   68
   13.4.5     453 Not Enough Bandwidth ............................   68
   13.4.6     454 Session Not Found ...............................   68
   13.4.7     455 Method Not Valid in This State ..................   68
   13.4.8     456 Header Field Not Valid for Resource .............   68
   13.4.9     457 Invalid Range ...................................   69
   13.4.10    458 Parameter Is Read-Only ..........................   69
   13.4.11    459 Aggregate Operation Not Allowed .................   69
   13.4.12    460 Only Aggregate Operation Allowed ................   69
   13.4.13    461 Unsupported Transport ...........................   69
   13.4.14    462 Destination Unreachable .........................   69
   13.4.15    470 Connection Authorization Required ...............   69
   13.4.16    471 Connection Credentials not accepted .............   69



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   13.5       Server Error 5xx ....................................   69
   13.5.1     551 Option not supported ............................   70
   14         Header Field Definitions ............................   70
   14.1       Accept ..............................................   72
   14.2       Accept-Credentials ..................................   72
   14.3       Accept-Encoding .....................................   77
   14.4       Accept-Language .....................................   77
   14.5       Accept-Ranges .......................................   77
   14.6       Allow ...............................................   77
   14.7       Authorization .......................................   77
   14.8       Bandwidth ...........................................   78
   14.9       Blocksize ...........................................   78
   14.10      Cache-Control .......................................   78
   14.11      Connection ..........................................   81
   14.12      Content-Base ........................................   81
   14.13      Content-Encoding ....................................   81
   14.14      Content-Language ....................................   81
   14.15      Content-Length ......................................   81
   14.16      Content-Location ....................................   81
   14.17      Content-Type ........................................   81
   14.18      CSeq ................................................   82
   14.19      Date ................................................   82
   14.20      ETag ................................................   82
   14.21      Expires .............................................   83
   14.22      From ................................................   83
   14.23      Host ................................................   84
   14.24      If-Match ............................................   84
   14.25      If-Modified-Since ...................................   84
   14.26      If-None-Match .......................................   84
   14.27      Last-Modified .......................................   85
   14.28      Location ............................................   85
   14.29      Proxy-Authenticate ..................................   85
   14.30      Proxy-Require .......................................   85
   14.31      Public ..............................................   85
   14.32      Range ...............................................   86
   14.33      Referer .............................................   88
   14.34      Retry-After .........................................   88
   14.35      Require .............................................   88
   14.36      RTP-Info ............................................   89
   14.37      Scale ...............................................   91
   14.38      Speed ...............................................   92
   14.39      Server ..............................................   92
   14.40      Session .............................................   92
   14.41      Supported ...........................................   94
   14.42      Timestamp ...........................................   95
   14.43      Transport ...........................................   95
   14.44      Unsupported .........................................  101
   14.45      User-Agent ..........................................  102



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   14.46      Vary ................................................  102
   14.47      Via .................................................  102
   14.48      WWW-Authenticate ....................................  102
   15         Caching .............................................  102
   16         Examples ............................................  103
   16.1       Media on Demand (Unicast) ...........................  103
   16.2       Streaming of a Container file .......................  106
   16.3       Single Stream Container Files .......................  109
   16.4       Live Media Presentation Using Multicast .............  111
   16.5       Capability Negotiation ..............................  112
   17         Security Framework ..................................  113
   17.1       RTSP and HTTP Authentication ........................  113
   17.2       RTSP over TLS .......................................  114
   17.3       Security and Proxies ................................  114
   17.3.1     Accept-Credentials ..................................  115
   17.3.2     User approved TLS procedure .........................  116
   18         Syntax ..............................................  118
   18.1       Base Syntax .........................................  118
   18.2       RTSP Protocol Definition ............................  119
   18.2.1     Generic Protocol elements ...........................  119
   18.2.2     Message Syntax ......................................  120
   18.2.3     Header Syntax .......................................  124
   19         Security Considerations .............................  126
   20         IANA Considerations .................................  128
   20.1       Feature-tags ........................................  129
   20.1.1     Description .........................................  129
   20.1.2     Registering New Feature-tags with IANA ..............  129
   20.1.3     Registered entries ..................................  130
   20.2       RTSP Methods ........................................  130
   20.2.1     Description .........................................  130
   20.2.2     Registering New Methods with IANA ...................  130
   20.2.3     Registered Entries ..................................  131
   20.3       RTSP Status Codes ...................................  131
   20.3.1     Description .........................................  131
   20.3.2     Registering New Status Codes with IANA ..............  131
   20.3.3     Registered Entries ..................................  131
   20.4       RTSP Headers ........................................  131
   20.4.1     Description .........................................  131
   20.4.2     Registering New Headers with IANA ...................  131
   20.4.3     Registered entries ..................................  132
   20.5       Transport Header registries .........................  132
   20.5.1     Transport Protocols .................................  132
   20.5.2     Profile .............................................  133
   20.5.3     Lower Transport .....................................  133
   20.5.4     Transport modes .....................................  134
   20.6       Cache Directive Extensions ..........................  134
   20.7       Accept-Credentials policies .........................  135
   20.8       SDP attributes ......................................  135



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   A          RTSP Protocol State Machine .........................  136
   A.1        States ..............................................  137
   A.2        State variables .....................................  137
   A.3        Abbreviations .......................................  137
   A.4        State Tables ........................................  137
   B          Media Transport Alternatives ........................  141
   B.1        RTP .................................................  141
   B.1.1      AVP .................................................  141
   B.1.2      AVP/UDP .............................................  141
   B.1.3      AVP/TCP .............................................  143
   B.1.4      Handling NPT Jumps in the RTP Media Layer ...........  143
   B.1.5      Handling RTP Timestamps after PAUSE .................  145
   B.1.6      RTSP / RTP Integration ..............................  148
   B.1.7      Scaling with RTP ....................................  148
   B.1.8      Maintaining NPT synchronization with RTP
              timestamps ..........................................  148
   B.1.9      Continuous Audio ....................................  148
   B.1.10     Multiple Sources in an RTP Session ..................  148
   B.1.11     Usage of SSRCs and the RTCP BYE Message During a
              RTSP Session ........................................  148
   B.2        Future Additions ....................................  149
   C          Use of SDP for RTSP Session Descriptions ............  150
   C.1        Definitions .........................................  150
   C.1.1      Control URL .........................................  150
   C.1.2      Media Streams .......................................  151
   C.1.3      Payload Type(s) .....................................  152
   C.1.4      Format-Specific Parameters ..........................  152
   C.1.5      Range of Presentation ...............................  152
   C.1.6      Time of Availability ................................  153
   C.1.7      Connection Information ..............................  153
   C.1.8      Entity Tag ..........................................  153
   C.2        Aggregate Control Not Available .....................  154
   C.3        Aggregate Control Available .........................  155
   C.4        RTSP external SDP delivery ..........................  156
   D          Minimal RTSP implementation .........................  156
   D.1        Client ..............................................  156
   D.1.1      Basic Playback ......................................  157
   D.1.2      Authentication-enabled ..............................  157
   D.2        Server ..............................................  157
   D.2.1      Basic Playback ......................................  158
   D.2.2      Authentication-enabled ..............................  159
   E          Open Issues .........................................  159
   F          Changes .............................................  160
   G          Author Addresses ....................................  166
   H          Contributors ........................................  167
   I          Acknowledgements ....................................  167
   J          Normative References ................................  168
   K          Informative References ..............................  169



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1 Introduction

1.1 RTSP Specification Update

   This document is a draft to an update of RTSP, a proposed standard     |
   defined in RFC 2326  [1]. The goal the update is to progress RTSP to   |
   draft standard status. Many flaws have been identified in RTSP since   |
   its publication. While this draft tries to address these flaws, not    |
   all known issues have been resolved.  Appendix  F catalogs the issues  |
   that have already been addressed. Known open issues are listed in      |
   appendix  E.                                                           |

   The possibility of progressing RTSP to draft standard without          |
   republishing RTSP as a proposed standard depends on the changes        |
   necessary to make the protocol work.                                   |

   A list of bugs against the specification is available at               |
   "http://rtspspec.sourceforge.net". These bugs should be taken into     |
   account when reading this specification. Input on the unresolved bugs  |
   and other issues can be sent via e-mail to the MMUSIC WG's mailing     |
   list mmusic@ietf.org and the authors.

   Not all of the contents of RFC 2326 are part of this draft. In an      |
   attempt to prevent bloat, the specification has been reduced and       |
   split.  The content of this draft is the core specification of the
   protocol. It contains the general idea behind RTSP and the basic
   functionality necessary to establish an on-demand play-back session.
   It also contains the mechanisms for extending the protocol. Any other
   functionality will be published as extension documents. The Working    |
   group is currently working on:

        o NAT and FW traversal mechanisms for RTSP are described in a
          document called "How to make Real-Time Streaming Protocol
          (RTSP) traverse Network Address Translators (NAT) and interact
          with Firewalls." [24].

   There have also been discussion or proposals about the following
   extensions to RTSP:

        o Mute and Unmute Extension [25].

        o RTSP Stream Switching [26].

        o Live Streaming Relays [27].

        o Unreliable transport of RTSP messages (rtspu).

        o The Record functionality.



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        o A text body type with suitable syntax for basic parameters to
          be used in SET_PARAMETER, and GET_PARAMETER. Including IANA
          registry within the defined name space.

        o A RTSP MIB.

1.2 Purpose

   The Real-Time Streaming Protocol (RTSP) establishes and controls
   single or several time-synchronized streams of continuous media such
   as audio and video. Put simply, RTSP acts as a "network remote
   control" for multimedia servers.

   There is no notion of a RTSP connection in the protocol. Instead, a
   RTSP server maintains a session labelled by an identifier to
   associate groups of media streams and their states. A RTSP session is
   not tied to a transport-level connection such as a TCP connection.
   During a session, a client may open and close many reliable transport
   connections to the server to issue RTSP requests for that session.

   This memorandum describes the use of RTSP over a reliable connection
   based transport level protocol such as TCP. RTSP may be implemented
   over an unreliable connectionless transport protocol such as UDP.
   While nothing in RTSP precludes this, additional definition of this
   problem area must be handled as an extension to the core
   specification.


        The mechanisms of RTSP's operation over UDP were left out
        of this spec. because they were poorly defined in RFC 2326
        [1] and the tradeoff in size and complexity of this spec.
        for a small gain in a targeted problem space was not deemed
        justifiable.

   The set of streams to be controlled in a RTSP session is defined by a  |
   presentation description. This memorandum does not define a format
   for the presentation description. However appendix C defines how SDP
   [2] is used for this purpose. The streams controlled by RTSP may use
   RTP [3] for their data transport, but the operation of RTSP does not
   depend on the transport mechanism used to carry continuous media.      |
   RTSP is intentionally similar in syntax and operation to HTTP/1.1 [4]
   so that extension mechanisms to HTTP can in most cases also be added
   to RTSP. However, RTSP differs in a number of important aspects from
   HTTP:

        o RTSP introduces a number of new methods and has a different
          protocol identifier.




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        o RTSP has the notion of a session built into the protocol.

        o A RTSP server needs to maintain state by default in almost all
          cases, as opposed to the stateless nature of HTTP.

        o Both a RTSP server and client can issue requests.

        o Data is usually carried out-of-band by a different protocol.
          Session descriptions returned in a DESCRIBE response (see
          Section 11.2) and interleaving of RTP with RTSP over TCP are
          exceptions to this rule (see Section 12).

        o RTSP is defined to use ISO 10646 (UTF-8) rather than ISO
          8859-1, consistent with HTML internationalization efforts
          [28].

        o The Request-URL always contains the absolute URL. Because of
          backward compatibility with a historical blunder, HTTP/1.1 [4]
          carries only the absolute path in the request and puts the
          host name in a separate header field.


             This makes "virtual hosting" easier, where a single
             host with one IP address hosts several document trees.

   The protocol supports the following operations:

        Retrieval of media from media server: The client can either
             request a presentation description via RTSP DESCRIBE, HTTP
             or some other method. If the presentation is being
             multicast, the presentation description contains the
             multicast addresses and ports to be used for the continuous
             media. If the presentation is to be sent only to the client
             via unicast, the client provides the destination for
             security reasons.

        Invitation of a media server to a conference: A media server can
             be "invited" to join an existing conference to play back
             media into the presentation. This mode is useful for
             distributed teaching applications. Several parties in the
             conference may take turns "pushing the remote control
             buttons".

   RTSP requests may be handled by proxies, tunnels and caches as in
   HTTP/1.1 [4].

1.3 Notational Conventions




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   Since many of the definitions and syntax are identical to HTTP/1.1,
   this specification only points to the section where they are defined
   rather than copying it. For brevity, [HX.Y] is to be taken to refer
   to Section X.Y of the current HTTP/1.1 specification (RFC 2616 [4]).

   All the mechanisms specified in this document are described in both
   prose and the augmented Backus-Naur form (BNF) described in detail in
   RFC 2234 [5].

   Indented and smaller-type paragraphs are used to provide background    |
   and motivation. This is intended to give readers who were not          |
   involved with the formulation of the specification an understanding    |
   of why things are the way they are in RTSP.

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC 2119 [6].

   The word, unspecified, is used to indicate functionality or features   |
   that are not defined in this specification. Such functionality cannot  |
   be used in a standardized manner without further definition and        |
   review in an extension specification to RTSP.

1.4 Terminology

   Some of the terminology has been adopted from HTTP/1.1 [4]. Terms not
   listed here are defined as in HTTP/1.1.

        Aggregate control: The concept of controlling multiple streams
             using a single timeline, generally maintained by the
             server. A client, for example, uses aggregate control when
             it issues a single play or pause message to simultaneously
             control both the audio and video in a movie.

        Aggregate control URL: The URL used in a RTSP request to refer
             to and control an aggregated session. It normally, but not
             always, corresponds to the presentation URL specified in
             the session description. See Section  11.3 for more
             information.

        Conference: a multiparty, multimedia presentation, where "multi"
             implies greater than or equal to one.

        Client: The client requests media service from the media server.

        Connection: A transport layer virtual circuit established
             between two programs for the purpose of communication.




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        Container file: A file which may contain multiple media streams   |
             which often constitutes a presentation when played           |
             together. The concept of a container file is not embedded    |
             in the protocol.  However, RTSP servers may offer aggregate  |
             control on the media streams within these files.

        Continuous media: Data where there is a timing relationship
             between source and sink; that is, the sink must reproduce
             the timing relationship that existed at the source. The
             most common examples of continuous media are audio and
             motion video. Continuous media can be real-time
             (interactive), where there is a "tight" timing relationship
             between source and sink, or streaming (playback), where the
             relationship is less strict.

        Entity: The information transferred as the payload of a request
             or response. An entity consists of meta-information in the
             form of entity-header fields and content in the form of an
             entity-body, as described in Section 8.

        Feature-tag: A tag representing a certain set of functionality,
             i.e. a feature.

        Live: Normally used to describe a presentation or session with
             media coming from ongoing event. This generally results in
             that the session has a unbound or only loosely defined
             duration, and that no seek operations are possible.

        Media initialization: Datatype/codec specific initialization.
             This includes such things as clock rates, color tables,
             etc. Any transport-independent information which is
             required by a client for playback of a media stream occurs
             in the media initialization phase of stream setup.

        Media parameter: Parameter specific to a media type that may be
             changed before or during stream playback.

        Media server: The server providing playback services for one or
             more media streams. Different media streams within a
             presentation may originate from different media servers.  A  |
             media server may reside on the same host or on a different   |
             host from which the presentation is invoked.

        Media server indirection: Redirection of a media client to a
             different media server.

        (Media) stream: A single media instance, e.g., an audio stream
             or a video stream as well as a single whiteboard or shared



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             application group. When using RTP, a stream consists of all
             RTP and RTCP packets created by a source within an RTP
             session. This is equivalent to the definition of a DSM-CC
             stream([29]).

        Message: The basic unit of RTSP communication, consisting of a
             structured sequence of octets matching the syntax defined
             in Section 18 and transmitted over a connection or a         |
             connectionless transport.

        Non-Aggregated Control: Control of a single media stream.  Only
             possible in RTSP sessions with a single media.

        Participant: Member of a conference. A participant may be a
             machine, e.g., a playback server.

        Presentation: A set of one or more streams presented to the
             client as a complete media feed and described by a           |
             presentation description as defined below. In the RTSP       |
             context, this generally implies aggregate control over the   |
             streams, but does not necessarily have to.

        Presentation description: A presentation description contains
             information about one or more media streams within a
             presentation, such as the set of encodings, network
             addresses and information about the content. Other IETF
             protocols such as SDP (RFC 2327 [2]) use the term "session"
             for a presentation. The presentation description may take
             several different formats, including but not limited to the
             session description protocol format, SDP.                    |

        Response: A RTSP response. If an HTTP response is meant, that is
             indicated explicitly.

        Request: A RTSP request. If an HTTP request is meant, that is
             indicated explicitly.

        Request URL: The URL used in a request to indicate the resource   |
             on which the request shall be performed.


        RTSP agent: Refers to either a RTSP client, a RTSP server, or a   |
             RTSP Proxy. In this specification, there are many            |
             capabilities that are common to these three entities such    |
             as the capability to send requests or receive responses.     |
             This term will be used when describing functionality that    |
             is applicable to all three of these entities.




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        RTSP session: A stateful abstraction upon which the main control
             methods of RTSP operate. A RTSP session is a server entity;
             it is created, maintained and destroyed by the server. It
             is established by a RTSP server upon the completion of a
             successful SETUP request (when 200 OK response is sent) and
             is labelled by a session identifier at that time. The
             session exists until timed out by the server or explicitly
             removed by a TEARDOWN request. A RTSP session is a stateful  |
             entity; a RTSP server maintains an explicit session state
             machine (see Appendix A) where most state transitions are
             triggered by client requests. The existence of a session
             implies the existence of state about the session's media
             streams and their respective transport mechanisms. A given
             session can have zero or more media streams associated with
             it. A RTSP server uses the session to aggregate control
             over multiple media streams.

        Transport initialization: The negotiation of transport
             information (e.g., port numbers, transport protocols)
             between the client and the server.

        URI: Universal Resource Identifier, see RFC 2396 [13]. In RTSP
             the used URIs are as general rule in fact URL's as they
             gives an location for the resource. Therefore although RTSP
             URLs are a subset of URIs, they will be refered as URLs.

        URL: Universal Resource Locator, is an URI which identifies the
             resource through its primary access mechanism, rather than
             identifying the resource by name or by some other
             attribute(s) of that resource.

1.5 Protocol Properties

   RTSP has the following properties:

        Extendable: New methods and parameters can be easily added to
             RTSP.

        Easy to parse: RTSP can be parsed by standard HTTP or MIME
             parsers.

        Secure: RTSP re-uses web security mechanisms, either at the
             transport level (TLS, RFC 2246 [7]) or within the protocol
             itself. All HTTP authentication mechanisms such as basic
             (RFC 2616 [4]) and digest authentication (RFC 2617 [8]) are
             directly applicable.

        Transport-independent: RTSP does not preclude the use of an



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             unreliable datagram protocol (UDP) (RFC 768 [9]), a
             reliable datagram protocol (RDP, RFC 1151, not widely used
             [30]) as it would be possible to implement application-
             level reliability. The use of a connectionless datagram
             protocol such as UDP or RDP requires additional definition
             that may be provided as extensions to the core RTSP
             specification. The usage of the reliable stream protocol
             TCP (RFC 793 [10]) is what is currently defined as
             transport protocol of RTSP messages.

        Multi-server capable: Each media stream within a presentation
             can reside on a different server. The client automatically
             establishes several concurrent control sessions with the
             different media servers.  Media synchronization is
             performed at the transport level.

        Separation of stream control and conference initiation:  Stream
             control is divorced from inviting a media server to a
             conference. In particular, SIP [31] or H.323 [32] may be
             used to invite a server to a conference.

        Suitable for professional applications: RTSP supports frame-
             level accuracy through SMPTE time stamps to allow remote
             digital editing.

        Presentation description neutral: The protocol does not impose a
             particular presentation description or metafile format and
             can convey the type of format to be used. However, the
             presentation description must contain at least one RTSP
             URL.

        Proxy and firewall friendly: The protocol should be readily
             handled by both application and transport-layer (SOCKS
             [33]) firewalls. A firewall may need to understand the
             SETUP method to open a "hole" for the media stream.

        HTTP-friendly: Where sensible, RTSP reuses HTTP concepts, so
             that the existing infrastructure can be reused. This
             infrastructure includes PICS (Platform for Internet Content
             Selection [34,35]) for associating labels with content.
             However, RTSP does not just add methods to HTTP since the
             controlling continuous media requires server state in most
             cases.

        Appropriate server control: If a client can start a stream, it
             must be able to stop a stream. Servers should not start
             streaming to clients in such a way that clients cannot stop
             the stream.



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        Transport negotiation: The client can negotiate the transport
             method prior to actually needing to process a continuous
             media stream.


        An earlier requirement in RTSP was multi-client capability.
        However, it was determined that a better approach was to
        make sure that the protocol is easily extensible to the
        multi-client scenario. Stream identifiers can be used by
        several control streams, so that "passing the remote" would
        be possible. The protocol would not address how several
        clients negotiate access; this is left to either a "social
        protocol" or some other floor control mechanism.

1.6 Extending RTSP

   Since not all media servers have the same functionality, media
   servers by necessity will support different sets of requests. For
   example:

        o A server may not be capable of seeking (absolute positioning)
          if it is to support live events only.

        o Some servers may not support setting stream parameters and
          thus not support GET_PARAMETER and SET_PARAMETER.

   A server SHOULD implement all header fields described in Section 14.

   It is up to the creators of presentation descriptions not to ask the
   impossible of a server. This situation is similar in HTTP/1.1 [4],
   where the methods described in [H19.5] are not likely to be supported
   across all servers.

   RTSP can be extended in three ways, listed here in order of the
   magnitude of changes supported:

        o Existing methods can be extended with new parameters, e.g.
          headers, as long as these parameters can be safely ignored by
          the recipient. If the client needs negative acknowledgement     |
          when a method extension is not supported, a tag corresponding
          to the extension may be added in the Require: field (see
          Section 14.35).

        o New methods can be added. If the recipient of the message does
          not understand the request, it responds with error code 501
          (Not Implemented) and the sender should not attempt to use
          this method again. A client may also use the OPTIONS method to
          inquire about methods supported by the server. The server MUST



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          list the methods it supports using the Public response header.

        o A new version of the protocol can be defined, allowing almost
          all aspects (except the position of the protocol version
          number) to change.

   The basic capability discovery mechanism can be used to both discover
   support for a certain feature and to ensure that a feature is
   available when performing a request. For detailed explanation of this
   see chapter  10.

1.7 Overall Operation

   Each presentation and media stream is identified by an RTSP URL.  The
   overall presentation and the properties of the media the presentation
   is made up of are defined by a presentation description file, the
   format of which is outside the scope of this specification. The
   presentation description file may be obtained by the client using
   HTTP or other means such as email and may not necessarily be stored
   on the media server.

   For the purposes of this specification, a presentation description is
   assumed to describe one or more presentations, each of which
   maintains a common time axis. For simplicity of exposition and
   without loss of generality, it is assumed that the presentation
   description contains exactly one such presentation. A presentation
   may contain several media streams.

   The presentation description file contains a description of the media
   streams making up the presentation, including their encodings,
   language, and other parameters that enable the client to choose the
   most appropriate combination of media. In this presentation
   description, each media stream that is individually controllable by
   RTSP is identified by a RTSP URL, which points to the media server
   handling that particular media stream and names the stream stored on
   that server. Several media streams can be located on different
   servers; for example, audio and video streams can be split across
   servers for load sharing. The description also enumerates which
   transport methods the server is capable of.

   Besides the media parameters, the network destination address and
   port need to be determined. Several modes of operation can be
   distinguished:

        Unicast: The media is transmitted to the source of the RTSP
             request, with the port number chosen by the client.
             Alternatively, the media is transmitted on the same
             reliable stream as RTSP.



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        Multicast, server chooses address: The media server picks the
             multicast address and port. This is the typical case for a
             live or near-media-on-demand transmission.

        Multicast, client chooses address: If the server is to
             participate in an existing multicast conference, the
             multicast address, port and encryption key are given by the
             conference description, established by means outside the
             scope of this specification.

1.8 RTSP States

   RTSP controls a stream which may be sent via a separate protocol,
   independent of the control channel. For example, RTSP control may
   occur on a TCP connection while the data flows via UDP. Thus, data
   delivery continues even if no RTSP requests are received by the media
   server. Also, during its lifetime, a single media stream may be
   controlled by RTSP requests issued sequentially on different TCP
   connections. Therefore, the server needs to maintain "session state"
   to be able to correlate RTSP requests with a stream. The state
   transitions are described in Appendix A.

   Many methods in RTSP do not contribute to state. However, the
   following play a central role in defining the allocation and usage of
   stream resources on the server: SETUP, PLAY, PAUSE, REDIRECT, PING
   and TEARDOWN.

        SETUP: Causes the server to allocate resources for a stream and
             create a RTSP session.

        PLAY: Starts data transmission on a stream allocated via SETUP.

        PAUSE: Temporarily halts a stream without freeing server
             resources.

        REDIRECT: Indicates that the session should be moved to new
             server / location

        PING: Prevents the identified session from being timed out.

        TEARDOWN: Frees resources associated with the stream.  The RTSP
             session ceases to exist on the server.

   RTSP methods that contribute to state use the Session header field
   (Section 14.40) to identify the RTSP session whose state is being
   manipulated. The server generates session identifiers in response to
   SETUP requests (Section 11.3).




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1.9 Relationship with Other Protocols

   RTSP has some overlap in functionality with HTTP. It also may
   interact with HTTP in that the initial contact with streaming content
   is often to be made through a web page. The current protocol
   specification aims to allow different hand-off points between a web
   server and the media server implementing RTSP. For example, the
   presentation description can be retrieved using HTTP or RTSP, which
   reduces round trips in web-browser-based scenarios, yet also allows
   for stand alone RTSP servers and clients which do not rely on HTTP at
   all. However, RTSP differs fundamentally from HTTP in that most data
   delivery takes place out-of-band in a different protocol. HTTP is an
   asymmetric protocol where the client issues requests and the server
   responds. In RTSP, both the media client and media server can issue
   requests. RTSP requests are also stateful; they may set parameters
   and continue to control a media stream long after the request has
   been acknowledged.


        Re-using HTTP functionality has advantages in at least two
        areas, namely security and proxies. The requirements are
        very similar, so having the ability to adopt HTTP work on
        caches, proxies and authentication is valuable.

   RTSP assumes the existence of a presentation description format that
   can express both static and temporal properties of a presentation
   containing several media streams. Session Description Protocol (SDP)
   [2] is generally the format of choice; however, RTSP is not bound to
   it. For data delivery, most real-time media will use RTP as a
   transport protocol. While RTSP works well with RTP, it is not tied to
   RTP.

2 RTSP Use Cases


   This section describes some of the use cases for RTSP. They are        |
   listed in descending order of importance in regards to ensuring that   |
   all necessary functionality is present. This specification does only   |
   fully support usage of the two first. Also in these first two cases    |
   are there special cases that will not be supported without             |
   extensions, e.g. the redirection of media to another address than the  |
   controlling entity.                                                    |

2.1 On-demand Playback of Stored Content                                  |

   An RTSP capable server stores content suitable for being streamed to   |
   a client. A client desiring playback of any of the stored content      |
   uses RTSP to set up the media transport required for the desired       |



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   content. Then RTSP is used to initiate, halt and manipulate the        |
   transmission of the content. There are also requirement on being able  |
   to use RTSP to carry necessary description and synchronization         |
   information for the content.                                           |

   The above high level description can be broken down into a number of   |
   functionalities that RTSP needs to be capable of.                      |

        Presentation Description: The possibility to carry                |
             initialization information about the presentation            |
             (content), for example, which media codec(s) that are        |
             needed for the content. Other information that are           |
             important; how many media stream that the presentation       |
             contains; what transport protocols to use for the media      |
             streams; and identifiers for these media streams.  This      |
             information is required before setup of the content is       |
             possible. The information is also needed by the client to    |
             determine if it is capable at all to support the content.    |

             This information is not required to be sent using RTSP,      |
             instead other external protocols can be utilized to          |
             transport presentation descriptions. Two good examples are   |
             the use of HTTP  [4] or email to fetch or receive            |
             presentation descriptions like SDP  [2].                     |

        Setup: Performing setup of some or all of the media streams in a  |
             presentation. The setup itself consist of determining which  |
             protocols for media transport to use; the necessary          |
             parameters for the protocol, like addresses and ports.       |

        Control of Transmission: After the necessary media streams has    |
             been established the client can request the server to start  |
             transmitting the content. There is need to allow the client  |
             to arbitrary times start or stop the transmission of the     |
             content. There are also exist need to be able to start the   |
             transmission at an any point in the timeline of the          |
             presentation.                                                |

        Synchronization: For media transport protocols like RTP  [18] it  |
             might be beneficial to carry synchronization information     |
             within RTSP. Either due to the lack of inter media           |
             synchronization within the protocol itself, or the           |
             potential delay before the synchronization is established    |
             (which is the case for RTP when using RTCP).                 |

        Termination There is also need to be able to terminate the        |
             established contexts.                                        |




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   For this use cases there is a number of assumption about how it        |
   works. These are listed below:                                         |

        On-Demand content: The content available is stored at the server  |
             and can be accessed at any time during a time period when    |
             it is intended to be available.                              |

        Independent sessions: A server is capable of serving a number of  |
             clients simultaneously, including from the same piece of     |
             content at different points in that presentations time-      |
             line.                                                        |

        Unicast Transport: Content for each individual client is          |
             transmitted to them using unicast traffic.                   |

   It is also possible to redirect the media traffic to another           |
   destination than where the entity controlling traffic uses.  However   |
   allowing this without appropriate mechanisms for checking that the     |
   destination approves of this is a denial of service threat.            |

2.2 Unicast distribution of Live Content                                  |

   This use cases is not that different from the above on-demand content  |
   case (see section  2.1. The difference is really the restriction the   |
   content itself establish.  Live content is continuously distributed    |
   as it becomes available from a source, i.e. the main difference to     |
   on-demand is that one starts distributing content before the end of    |
   it has become available to the server.                                 |

   In many cases the consumer of live content is only interested in       |
   consuming what is actually happens "now", i.e. very similar to         |
   broadcast TV. However in this case it is assumed that there exist no   |
   broadcast or multicast channel to the users, and instead the server    |
   functions as a distribution node, sending the same content to          |
   multiple receivers, using unicast traffic between server and client.   |
   This unicast traffic and the transport parameters are individually     |
   negotiated for each receiving client.                                  |

   Another aspect of live content is that it has often very limited time  |
   of availability, as it is only is available for the duration of the    |
   event the content covers. A example of such a live content could for   |
   example be a music concert, which lasts 2 hour and starts at a         |
   predetermined time. Thus there is need to announce when and for how    |
   long the live content is available.                                    |

2.3 On-demand Playback using Multicast                                    |

   It is possible to use RTSP to request that media is delivered to a     |



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   multicast group. The entity setting up the session (the controller)    |
   will then control when and what media that is delivered to the group.  |
   Also this use case has some potential for denial of service attacks,   |
   in this case flooding any multicast group. Therefore there is need     |
   for a mechanism indicating that the group actually accepts the         |
   traffic from the RTSP server.                                          |

   An open issue in this use case is how one ensures that all receivers   |
   listening to the multicast or broadcast receives the session           |
   presentation configuring the receivers.                                |

2.4 Inviting a RTSP server into a conference                              |

   If one has an established conference or group session, it is possible  |
   to have a RTSP server distribute media to the whole group. The         |
   transmission to the group is simplest controlled by a single           |
   participant or leader of the conference. Shared control might be       |
   possible, but would require further investigation and possibly         |
   extensions. There are some protocol mechanisms missing for this        |
   scenario.                                                              |

   For reasonable complexity in the media transmission stage, this use    |
   case assumes that there exist either multicast or a conference focus   |
   that redistribute media to all participants.                           |

   In some more detail, this use case is intended to be able to handle    |
   the following scenario: A conference leader or participant (from here  |
   called the controller) has some pre-stored content on a RTSP server    |
   that he likes to share with the group. The controller sets up a RTSP   |
   session at the streaming server for the content the controller likes   |
   to share. The session description for the content is retrieved to the  |
   controller. The media destination for the media content is set to the  |
   shared multicast group or conference focus. When desired by the        |
   controller, he/she can start and stop the transmission of the media    |
   to the conference group.                                               |

   There are several issues with this use case that is not solved by      |
   this core specification for RTSP:                                      |

        o Denial of service threat, to avoid a RTSP server from being a   |
          unknowing participant of a denial of service attack the server  |
          must be able to verify the destinations acceptance for the      |
          media. Such a mechanism does not yet exist that can be used to  |
          verify the approval to received media, instead only policies    |
          can be used, which can be made to work in controlled            |
          environments.                                                   |

        o The problem of distributing the presentation description to     |



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          all participants in the group. To enable a media receiver to    |
          decode the content correctly the media configuration            |
          information will need to be distributed reliable to all         |
          participants. This will most likely require support from an     |
          external protocol.                                              |

        o Passing the control. If it is desired to be able to pass the    |
          control of the RTSP session between the participants some       |
          support will be required by an external protocol for the        |
          necessary exchange of state information and possibly floor      |
          control of who is controlling the RTSP session.                 |

   So if there interest in this use case further work on the necessary    |
   extensions has to be performed.                                        |

2.5 Live Content using Multicast                                          |

   This use case does in its simplest form not require any use of RTSP    |
   at all. This is what multicast conferences being announce with SAP     |
   and SDP are intended to handle. However in use cases where more        |
   advance features like access control to the multicast session is       |
   desired, RTSP could be used for session establishment.                 |

   A client desiring to join a live multicasted media session with        |
   cryptographic (encryption) access control could use RTSP in the        |
   following way. The source of the session, announces the session and    |
   gives all interested to join, a RTSP URI. The client connects to the   |
   server and requests the presentation description allowing for          |
   configuration the reception. In this step it is possible to use        |
   secured transport for the client, and also desired levels of           |
   authentication, for example for charging purposes or simply access     |
   control. An RTSP link also allows for load balancing between multiple  |
   servers. However if this the only thing that occurs it can probably    |
   be solved as simple using HTTP.                                        |

   However for session where the sender likes to keep track of each       |
   individual receiver during the session, and possibly use this side     |
   channel for pushing out key-updates or other side information that is  |
   desirable to be done on a per receiver basis, and the receivers are    |
   not know prior to the session start, the state establishment that      |
   RTSP provides can be beneficial. In this case a client would           |
   establish a RTSP session to the multicast group. The RTSP server will  |
   not transmit any media, instead it will simply point to the multicast  |
   group. However the client and server will be able to keep the session  |
   alive for as long as the receiver participates in the session. Thus    |
   enabling for example server to client pushes of updates.               |

   This use cases will most likely not be able to actually implement      |



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   some extensions in relation to the server to client push mechanism.    |
   Here a method like ANNOUNCE might be suitable, however it will         |
   require a RTSP extension to revive the method.

3 Protocol Parameters

3.1 RTSP Version

   HTTP Specification Section [H3.1] applies, with HTTP replaced by
   RTSP. This specification defines version 1.0 of RTSP.

3.2 RTSP URL

   The "rtsp", "rtsps" and "rtspu" schemes are used to refer to network
   resources via the RTSP protocol. This section defines the scheme-
   specific syntax and semantics for RTSP URLs. The RTSP URL is case
   sensitive.

   Informative RTSP URL syntax:

   rtsp[u|s]://host[:port]/abspath[?query]#fragment


   See section  18.2.1 for the formal definition of the RTSP URL syntax.


        Note that fragment and query identifiers do not have a
        well-defined meaning at this time, i.e. their usage is
        unspecified, with the interpretation left to the RTSP
        server.

   The URL scheme rtsp requires that commands are issued via a reliable   |
   protocol (within the Internet, TCP), while the scheme rtspu is         |
   intended to identify RTSP over an unreliable protocol (within the      |
   Internet, UDP). The scheme rtsps identifies a reliable transport       |
   using secure transport (TLS [7]). The rtspu is not defined in this     |
   specification, and are for future extensions of the protocol to        |
   define how to use.

   If the port is empty or not given, port 554 SHALL be assumed. The
   semantics are that the identified resource can be controlled by RTSP
   at the server listening for TCP (scheme "rtsp") connections or UDP
   (scheme "rtspu") packets on that port of host, and the Request-URL
   for the resource is rtsp_URL. For the scheme rtsps the TCP and UDP
   port 322 is registered and SHALL be assumed.

   The use of IP addresses in URLs SHOULD be avoided whenever possible
   (see RFC 1924 [11]). Note: Using qualified domain names in any URL is



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   one requirement for making it possible for RFC 2326 implementations
   of RTSP to use IPv6. This specification is updated to allow for
   literal IPv6 addresses in RTSP URLs using the host specification in
   RFC 2732 [12].

   A presentation or a stream is identified by a textual media
   identifier, using the character set and escape conventions [H3.2] of
   URLs (RFC 2396 [13]). URLs may refer to a stream or an aggregate of
   streams, i.e., a presentation. Accordingly, requests described in
   Section 11 can apply to either the whole presentation or an
   individual stream within the presentation. Note that some request
   methods can only be applied to streams, not presentations and vice
   versa.

   For example, the RTSP URL:

     rtsp://media.example.com:554/twister/audiotrack


   identifies the audio stream within the presentation "twister", which
   can be controlled via RTSP requests issued over a TCP connection to
   port 554 of host media.example.com

   Also, the RTSP URL:

     rtsp://media.example.com:554/twister


   identifies the presentation "twister", which may be composed of audio
   and video streams.


        This does not imply a standard way to reference streams in
        URLs. The presentation description defines the hierarchical
        relationships in the presentation and the URLs for the
        individual streams. A presentation description may name a
        stream "a.mov" and the whole presentation "b.mov".

   The path components of the RTSP URL are opaque to the client and do
   not imply any particular file system structure for the server.


        This decoupling also allows presentation descriptions to be
        used with non-RTSP media control protocols simply by
        replacing the scheme in the URL.

3.3 Session Identifiers




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   Session identifiers are strings of any arbitrary length. A session
   identifier MUST be chosen randomly and MUST be at least eight
   characters long to make guessing it more difficult. (See Section 19.)

3.4 SMPTE Relative Timestamps

   A SMPTE relative timestamp expresses time relative to the start of
   the clip. Relative timestamps are expressed as SMPTE time codes for
   frame-level access accuracy. The time code has the format
                  hours:minutes:seconds:frames.subframes,
   with the origin at the start of the clip. The default smpte format
   is"SMPTE 30 drop" format, with frame rate is 29.97 frames per second.
   Other SMPTE codes MAY be supported (such as "SMPTE 25") through the
   use of alternative use of "smpte time". For the "frames" field in the
   time value can assume the values 0 through 29. The difference between
   30 and 29.97 frames per second is handled by dropping the first two
   frame indices (values 00 and 01) of every minute, except every tenth
   minute. If the frame value is zero, it may be omitted. Subframes are
   measured in one-hundredth of a frame.

   Examples:

     smpte=10:12:33:20-
     smpte=10:07:33-
     smpte=10:07:00-10:07:33:05.01
     smpte-25=10:07:00-10:07:33:05.01



3.5 Normal Play Time

   Normal play time (NPT) indicates the stream absolute position
   relative to the beginning of the presentation, not to be confused
   with the Network Time Protocol (NTP). The timestamp consists of a
   decimal fraction. The part left of the decimal may be expressed in
   either seconds or hours, minutes, and seconds. The part right of the
   decimal point measures fractions of a second.

   The beginning of a presentation corresponds to 0.0 seconds.  Negative
   values are not defined. The special constant now is defined as the
   current instant of a live type event. It MAY only be used for live
   type events, and SHALL NOT be used for on-demand content.

   NPT is defined as in DSM-CC: "Intuitively, NPT is the clock the
   viewer associates with a program. It is often digitally displayed on
   a VCR. NPT advances normally when in normal play mode (scale = 1),
   advances at a faster rate when in fast scan forward (high positive
   scale ratio), decrements when in scan reverse (high negative scale



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   ratio) and is fixed in pause mode. NPT is (logically) equivalent to
   SMPTE time codes." [29]

   Examples:

     npt=123.45-125
     npt=12:05:35.3-
     npt=now-




        The syntax conforms to ISO 8601. The npt-sec notation is
        optimized for automatic generation, the ntp-hhmmss notation
        for consumption by human readers. The "now" constant allows
        clients to request to receive the live feed rather than the
        stored or time-delayed version. This is needed since
        neither absolute time nor zero time are appropriate for
        this case.

3.6 Absolute Time

   Absolute time is expressed as ISO 8601 timestamps, using UTC (GMT).
   Fractions of a second may be indicated.

   Example for November 8, 1996 at 14h37 and 20 and a quarter seconds
   UTC:

     19961108T143720.25Z



3.7 Feature-tags

   Feature-tags are unique identifiers used to designate features in
   RTSP. These tags are used in Require (Section 14.35), Proxy-Require
   (Section 14.30), Unsupported (Section 14.44), and Supported (Section
   14.41) header fields.

   Feature tag needs to indicate if they apply to servers only, proxies
   only, or both server and proxies.

   The creator of a new RTSP feature-tag should either prefix the
   feature-tag with a reverse domain name (e.g.,
   "com.example.mynewfeature" is an apt name for a feature whose
   inventor can be reached at "example.com"), or register the new
   feature-tag with the Internet Assigned Numbers Authority (IANA), see
   IANA Section  20.



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   The usage of feature tags are further described in section 10 that     |
   deals with capability handling.                                        |

3.8 Entity Tags                                                           |

   Entity tags opaque strings that are used to compare two entities from  |
   the same resource, for example in caches or to optimize setup after a  |
   redirect. Further explanation is present in [H3.11]. For explanation   |
   on how to compare Entity tags see [H13.3]. Entity tags can be carried  |
   in the ETag header (see section 14.20) or in SDP (see section C.1.8).  |

   Entity tags are used in RTSP to make some methods conditional. The     |
   methods are made conditional through the inclusion of headers, see     |
   14.24 and 14.26.

4 RTSP Message

   RTSP is a text-based protocol and uses the ISO 10646 character set in
   UTF-8 encoding (RFC 2279 [14]). Lines are terminated by CRLF, but
   receivers should be prepared to also interpret CR and LF by
   themselves as line terminators.


        Text-based protocols make it easier to add optional
        parameters in a self-describing manner. Since the number of
        parameters and the frequency of commands is low, processing
        efficiency is not a concern. Text-based protocols, if done
        carefully, also allow easy implementation of research
        prototypes in scripting languages such as Tcl, Visual Basic
        and Perl.

   The 10646 character set avoids tricky character set switching, but is
   invisible to the application as long as US-ASCII is being used.  This
   is also the encoding used for RTCP. ISO 8859-1 translates directly
   into Unicode with a high-order octet of zero. ISO 8859-1 characters
   with the most-significant bit set are represented as 1100001x
   10xxxxxx. (See RFC 2279 [14])

   RTSP messages can be carried over any lower-layer transport protocol
   that is 8-bit clean. RTSP messages are vulnerable to bit errors and
   SHOULD NOT be subjected to them.

   Requests contain methods, the object the method is operating upon and
   parameters to further describe the method. Methods are idempotent,
   unless otherwise noted. Methods are also designed to require little
   or no state maintenance at the media server.

4.1 Message Types



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   See [H4.1].

4.2 Message Headers

   See [H4.2].

4.3 Message Body

   See [H4.3]

4.4 Message Length

   When a message body is included with a message, the length of that
   body is determined by one of the following (in order of precedence):

        1.   Any response message which MUST NOT include a message body
             (such as the 1xx, 204, and 304 responses) is always
             terminated by the first empty line after the header fields,
             regardless of the entity-header fields present in the
             message. (Note: An empty line consists of only CRLF.)

        2.   If a Content-Length header field (section 14.15) is
             present, its value in bytes represents the length of the
             message-body. If this header field is not present, a value
             of zero is assumed.

   Note that RTSP does not (at present) support the HTTP/1.1 "chunked"
   transfer coding(see [H3.6.1]) and requires the presence of the
   Content-Length header field.


        Given the moderate length of presentation descriptions
        returned, the server should always be able to determine its
        length, even if it is generated dynamically, making the
        chunked transfer encoding unnecessary.

5 General Header Fields

   See [H4.5], except that Pragma, Trailer, Transfer-Encoding, Upgrade,
   and Warning headers are not defined. RTSP further defines the CSeq,
   and Timestamp. The general headers are listed in table 1:


6 Request

   A request message from a client to a server or vice versa includes,    |
   within the first line (Request Line) of that message, the method to    |
   be applied to the resource, the identifier of the resource, and the    |



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                     Header Name    Comment
                     _________________________________
                     Cache-Control  See section 14.10
                     Connection     See section 14.11
                     CSeq           See section 14.18
                     Date           See section 14.19
                     Supported      See section 14.41
                     Timestamp      See section 14.42
                     Via            See section 14.47


   Table 1: The General headers used in RTSP.

   protocol version in use. Then follows zero or more headers that can    |
   be of general (Section 5), request (Section 6.2), or entity (Section   |
   8.1) type. Then an empty line, i.e. a line with only the two           |
   characters Carriage Return (CR) and Line Feed (LF), indicates the end  |
   of the header part. Optionally a message body (entity) follows to the  |
   end of the message. The length of the message body is indicated        |
   through the Content-Length entity header.

6.1 Request Line

   The request line, provides the most important things about the
   request: What method, on what resources and using which RTSP version.
   The methods that is defined by this specification can be seen in
   Table  6.1. The resource is identified through an absolute RTSP URL
   (see section 3.2.


   <Method> SP <Request-URL> SP <RTSP-Version> CRLF


   Please note: The request line's syntax can't be freely changed in      |
   future versions of RTSP, as this line indicates the version of the     |
   messages and need to be parsable also by older versions.

   Note that in contrast to HTTP/1.1 [4], RTSP requests always contain
   the absolute URL (that is, including the scheme, host and port)
   rather than just the absolute path.


        HTTP/1.1 requires servers to understand the absolute URL,
        but clients are supposed to use the Host request header.
        This is purely needed for backward-compatibility with
        HTTP/1.0 servers, a consideration that does not apply to
        RTSP.



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                     Method         Defined In Section
                     _________________________________
                     DESCRIBE       Section 11.2
                     GET_PARAMETER  Section 11.7
                     OPTIONS        Section 11.1
                     PAUSE          Section 11.5
                     PLAY           Section 11.4
                     PING           Section 11.10
                     REDIRECT       Section 11.9
                     SETUP          Section 11.3
                     SET_PARAMETER  Section 11.8
                     TEARDOWN       Section 11.6


   Table 2: The RTSP Methods


   The asterisk "*" in the Request-URL means that the request does not
   apply to a particular resource, but to the server or proxy itself,
   and is only allowed when the method used does not necessarily apply
   to a resource.

   One example would be as follows:


     OPTIONS * RTSP/1.0



   An OPTIONS in this form will determine the capabilities of the server
   or the proxy that first receives the request. If one needs to address
   the server explicitly, then one should use an absolute URL with the
   server's address.


     OPTIONS rtsp://example.com RTSP/1.0



6.2 Request Header Fields

   The RTSP headers in Table 3 can be included in a request with the
   purpose to give further define how the request should be fulfilled. A
   request header MAY also be response header, see section  7.1.2.


7 Response



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                   Header             Defined in Section
                   _____________________________________
                   Accept             Section 14.1
                   Accept-Encoding    Section 14.3
                   Accept-Language    Section 14.4
                   Authorization      Section 14.7
                   Bandwidth          Section 14.8
                   Blocksize          Section 14.9
                   From               Section 14.22
                   If-Match           Section 14.24
                   If-Modified-Since  Section 14.25
                   If-None-Match      Section 14.26
                   Proxy-Require      Section 14.30
                   Range              Section 14.32
                   Referer            Section 14.33
                   Require            Section 14.35
                   Scale              Section 14.37
                   Session            Section 14.40
                   Speed              Section 14.38
                   Supported          Section 14.41
                   Transport          Section 14.43
                   User-Agent         Section 14.45


   Table 3: The RTSP request headers

   [H6] applies except that HTTP-Version is replaced by RTSP-Version.
   Also, RTSP defines additional status codes and does not define some
   HTTP codes. The valid response codes and the methods they can be used
   with are defined in Table 4.

   After receiving and interpreting a request message, the recipient
   responds with an RTSP response message.

7.1 Status-Line

   The first line of a Response message is the Status-Line, consisting
   of the protocol version followed by a numeric status code, and the
   textual phrase associated with the status code, with each element
   separated by SP characters. No CR or LF is allowed except in the
   final CRLF sequence.


   <RTSP-Version> SP <Status-Code> SP <Reason-Phrase> CRLF


7.1.1 Status Code and Reason Phrase



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   The Status-Code element is a 3-digit integer result code of the
   attempt to understand and satisfy the request. These codes are fully
   defined in Section 13. The Reason-Phrase is intended to give a short
   textual description of the Status-Code. The Status-Code is intended
   for use by automata and the Reason-Phrase is intended for the human
   user. The client is not required to examine or display the Reason-
   Phrase.

   The first digit of the Status-Code defines the class of response. The
   last two digits do not have any categorization role.  There are 5
   values for the first digit:

        o 1xx: Informational - Request received, continuing process

        o 2xx: Success - The action was successfully received,
          understood, and accepted

        o 3rr: Redirection - Further action must be taken in order to
          complete the request

        o 4xx: Client Error - The request contains bad syntax or cannot
          be fulfilled

        o 5xx: Server Error - The server failed to fulfill an apparently
          valid request

   The individual values of the numeric status codes defined for
   RTSP/1.0, and an example set of corresponding Reason-Phrase's, are
   presented in table  4. The reason phrases listed here are only
   recommended they may be replaced by local equivalents without
   affecting the protocol. Note that RTSP adopts most HTTP/1.1 [4]
   status codes and adds RTSP-specific status codes starting at x50 to
   avoid conflicts with newly defined HTTP status codes.

   RTSP status codes are extensible. RTSP applications are not required
   to understand the meaning of all registered status codes, though such
   understanding is obviously desirable. However, applications MUST
   understand the class of any status code, as indicated by the first
   digit, and treat any unrecognized response as being equivalent to the
   x00 status code of that class, with the exception that an
   unrecognized response MUST NOT be cached. For example, if an
   unrecognized status code of 431 is received by the client, it can
   safely assume that there was something wrong with its request and
   treat the response as if it had received a 400 status code. In such
   cases, user agents SHOULD present to the user the entity returned
   with the response, since that entity is likely to include human-
   readable information which will explain the unusual status.




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7.1.2 Response Header Fields

   The response-header fields allow the request recipient to pass
   additional information about the response which cannot be placed in
   the Status-Line. These header fields give information about the
   server and about further access to the resource identified by the
   Request-URL. All headers currently being classified as response
   headers are listed in table  5.


   Response-header field names can be extended reliably only in
   combination with a change in the protocol version. However, new or
   experimental header fields MAY be given the semantics of response-
   header fields if all parties in the communication recognize them to
   be response-header fields. Unrecognized header fields are treated as
   entity-header fields.

8 Entity

   Request and Response messages MAY transfer an entity if not otherwise
   restricted by the request method or response status code. An entity
   consists of entity-header fields and an entity-body, although some
   responses will only include the entity-headers.

   The SET_PARAMETER, and GET_PARAMETER request and response, and
   DESCRIBE response MAY have an entity. All 4xx and 5xx responses MAY
   also have an entity.

   In this section, both sender and recipient refer to either the client
   or the server, depending on who sends and who receives the entity.

8.1 Entity Header Fields

   Entity-header fields define optional meta-information about the
   entity-body or, if no body is present, about the resource identified
   by the request. The entity header fields are listed in table  8.1.


   The extension-header mechanism allows additional entity-header fields
   to be defined without changing the protocol, but these fields cannot
   be assumed to be recognizable by the recipient. Unrecognized header
   fields SHOULD be ignored by the recipient and forwarded by proxies.

8.2 Entity Body

   See [H7.2] with the addition that a RTSP message with an entity body
   MUST include the Content-Type and Content-Length headers.



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9 Connections

   RTSP requests can be transmitted in several different ways:

        o persistent transport connections used for several request-
          response transactions;

        o one connection per request/response transaction;

        o connectionless mode.

   The type of transport is defined by the RTSP URL (Section 3.2). For
   the scheme "rtsp", a connection is assumed, while the scheme "rtspu"
   calls for RTSP requests to be sent without setting up a connection.

   Unlike HTTP, RTSP allows the media server to send requests to the
   media client. However, this is only supported for persistent
   connections, as the media server otherwise has no reliable way of
   reaching the client.  Also, this is the only way that requests from
   media server to client are likely to traverse firewalls.

9.1 Pipelining

   A client that supports persistent connections or connectionless mode
   MAY "pipeline" its requests (i.e., send multiple requests without
   waiting for each response). A server MUST send its responses to those
   requests in the same order that the requests were received.

9.2 Reliability and Acknowledgements

   The transmission of RTSP over UDP was optionally to implement and
   specified in RFC 2326. However that definition was not satisfactory
   for interoperable implementations. Due to lack of interest, this
   specification does not specify how RTSP over UDP shall be
   implemented. However to maintain backwards compatibility in the
   message format certain RTSP headers must be maintained.  These
   mechanism are described below. The next section Unreliable Transport
   (section  9.3) provides documentation of certain features that are
   necessary for transport protocols like UDP.

   Any RTSP request according to this specification SHALL NOT be sent to
   a multicast address. Any RTSP request SHALL be acknowledged. If a
   reliable transport protocol is used to carry RTSP, requests MUST NOT
   be retransmitted; the RTSP application MUST instead rely on the
   underlying transport to provide reliability.


        If both the underlying reliable transport such as TCP and



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        Code  Reason                               Method
        __________________________________________________________
        100   Continue                             all

        __________________________________________________________
        200   OK                                   all
        201   Created                              RECORD
        250   Low on Storage Space                 RECORD
        __________________________________________________________
        300   Multiple Choices                     all
        301   Moved Permanently                    all
        302   Found                                all
        303   See Other                            all
        305   Use Proxy                            all

        __________________________________________________________
        400   Bad Request                          all
        401   Unauthorized                         all
        402   Payment Required                     all
        403   Forbidden                            all
        404   Not Found                            all
        405   Method Not Allowed                   all
        406   Not Acceptable                       all
        407   Proxy Authentication Required        all
        408   Request Timeout                      all
        410   Gone                                 all
        411   Length Required                      all
        412   Precondition Failed                  DESCRIBE, SETUP
        413   Request Entity Too Large             all
        414   Request-URL Too Long                 all
        415   Unsupported Media Type               all
        451   Parameter Not Understood             SET_PARAMETER
        452   reserved                             n/a
        453   Not Enough Bandwidth                 SETUP
        454   Session Not Found                    all
        455   Method Not Valid In This State       all
        456   Header Field Not Valid               all
        457   Invalid Range                        PLAY, PAUSE
        458   Parameter Is Read-Only               SET_PARAMETER
        459   Aggregate Operation Not Allowed      all
        460   Only Aggregate Operation Allowed     all
        461   Unsupported Transport                all
        462   Destination Unreachable              all
        470   Connection Authorization Required    all
        471   Connection Credentials not accepted  all
        __________________________________________________________
        500   Internal Server Error                all
        501   Not Implemented                      all
        502   Bad Gateway                          all
        503   Service Unavailable                  all
        504   Gateway Timeout                      all
        505   RTSP Version Not Supported           all


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   Table 4: Status codes and their usage with RTSP methods


                  Header              Defined in Section
                  ______________________________________
                  Accept-Ranges       Section  14.5
                  ETag                Section  14.20
                  Location            Section  14.28
                  Proxy-Authenticate  Section  14.29
                  Public              Section  14.31
                  Range               Section  14.32
                  Retry-After         Section  14.34
                  RTP-Info            Section  14.36
                  Scale               Section  14.37
                  Session             Section  14.40
                  Server              Section  14.39
                  Speed               Section  14.38
                  Transport           Section  14.43
                  Unsupported         Section  14.44
                  Vary                Section  14.46
                  WWW-Authenticate    Section  14.48


   Table 5: The RTSP response headers


                   Header            Defined in Section

                   ____________________________________
                   Allow             Section  14.6
                   Content-Base      Section  14.12
                   Content-Encoding  Section  14.13
                   Content-Language  Section  14.14
                   Content-Length    Section  14.15
                   Content-Location  Section  14.16
                   Content-Type      Section  14.17
                   Expires           Section  14.21
                   Last-Modified     Section  14.27


   Table 6: The RTSP entity headers

        the RTSP application retransmit requests, it is possible
        that each packet loss results in two retransmissions. The
        receiver cannot typically take advantage of the
        application-layer retransmission since the transport stack
        will not deliver the application-layer retransmission
        before the first attempt has reached the receiver. If the
        packet loss is caused by congestion, multiple
        retransmissions at different layers will exacerbate the



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        congestion.

   Each request carries a sequence number in the CSeq header (Section
   14.18), which MUST be incremented by one for each distinct request
   transmitted to the destination end-point.  The initial sequence
   number MAY be chosen arbitrary, but is RECOMMENDED to begin with 0.
   If a request is repeated because of lack of acknowledgement, the
   request MUST carry the original sequence number (i.e., the sequence
   number is not incremented).

9.3 Unreliable Transport

   This section provides some information to future specification of
   RTSP over unreliable transport.

   Requests shall be acknowledged by the receiver. If there is no
   acknowledgement, the sender may resend the same message after a
   timeout of one round-trip time (RTT). The round-trip time is
   estimated as in TCP (RFC 1123) [15], with an initial round-trip value
   of 500 ms. An implementation MAY cache the last RTT measurement as
   the initial value for future connections.

   If RTSP is used over a small-RTT LAN, standard procedures for
   optimizing initial TCP round trip estimates, such as those used in
   T/TCP (RFC 1644) [36], can be beneficial.

   The Timestamp header (Section 14.42) is used to avoid the
   retransmission ambiguity problem [37] and obviates the need for
   Karn's algorithm.

   If a request is repeated because of lack of acknowledgement, the
   request must carry the original sequence number (i.e., the sequence
   number is not incremented).

   A number of RTSP messages destined for the same control end point may
   be packed into a single lower-layer PDU.

   The default port for the RTSP server is 554 for UDP.

9.4 The usage of connections

   Systems implementing RTSP MUST support carrying RTSP over TCP.  The
   default port for the RTSP server is 554 for TCP. A number of RTSP
   packets destined for the same control end point may be encapsulated
   into a TCP stream. RTSP data MAY be interleaved with RTP and RTCP
   packets, see section  12. Unlike HTTP, an RTSP message MUST contain a
   Content-Length header field whenever that message contains a payload
   (entity).  Otherwise, an RTSP packet is terminated with an empty line



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   immediately following the last message header.

   TCP can be used for both persistent connections and for one message
   exchange per connection, as presented above. This section gives
   further rules and recommendations on how to handle these connections
   so maximum interoperability and flexibility can be achieved.

   A server SHALL handle both persistent connections and one
   request/response transaction per connection. A persistent connection
   MAY be used for all transactions between the server and client,
   including messages to multiple RTSP sessions. However the persistent
   connection MAY also be closed after a few message exchanges, e.g. the
   initial setup and play command in a session. Later when the client
   wishes to send a new request, e.g.  pause, to the session a new
   connection is opened. This connection may either be for a single
   message exchange or can be kept open for several messages, i.e.
   persistent.

   A major motivation for allowing non-persistent connections are that
   they ensure fault tolerance. A second one is to allow for application
   layer mobility. A server and client supporting non-persistent
   connection can survive a loss of a TCP connection, e.g. due to a NAT
   timeout. When the client has discovered that the TCP connection has
   been lost, it can set up a new one when there is need to communicate.

   The client MAY close the connection at any time when no outstanding
   request/response transactions exist. The server SHOULD NOT close the
   connection unless at least one RTSP session timeout period has passed
   without data traffic. A server SHOULD NOT initiate a close of a
   connection directly after responding to a TEARDOWN request for the
   whole session. A server SHOULD NOT close the connection as a result
   of responding to a request with an error code. Doing this would
   prevent or result in extra overhead for the client when testing
   advanced or special types of requests.

   The client SHOULD NOT have more than one connection to the server at
   any given point. If a client or proxy handles multiple RTSP sessions
   on the same server, it is RECOMMENDED to use only a single
   connection.

   Older services which was implemented according to RFC 2326 sometimes
   requires the client to use persistent connection. The client closing
   the connection may result in that the server removes the session. To
   achieve interoperability with old servers any client is strongly
   RECOMMENDED to use persistent connections.

   A Client is also strongly RECOMMENDED to use persistent connections
   as it allows the server to send request to the client.  In cases



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   where no connection exist between the server and the client, this may
   cause the server to be forced to drop the RTSP session without
   notifying the client why, due to the lack of signalling channel. An
   example of such a case is when the server desires to send a REDIRECT
   request for a RTSP session to the client.

   A server implemented according to this specification MUST respond
   that it supports the "play.basic" feature-tag above. A client MAY
   send a request including the Supported header in a request to
   determine support of non-persistent connections. A server supporting
   non-persistent connections will return the "play.basic" feature-tag
   in its response. If the client receives the feature-tag in the
   response, it can be certain that the server handles non-persistent
   connections.

9.5 Timing Out RTSP messages

   Receivers of a request (responder) SHOULD respond to requests in a
   timely manner even when a reliable transport such as TCP is used.
   Similarly, the sender of a request (requestor) SHOULD wait for a
   sufficient time for a response before concluding that the responder
   will not be acting upon its request.

   A responder SHOULD respond to all requests within 5 seconds. If the
   responder recognizes that processing of a request will take longer
   than 5 seconds, it SHOULD send a 100 response as soon as possible. It
   SHOULD continue sending a 100 response every 5 seconds thereafter
   until it is ready to send the final response to the requestor. After
   sending a 100 response, the receiver MUST send a final response
   indicating the success or failure of the request.

   A requestor SHOULD wait at least 10 seconds for a response before
   concluding that the responder will not be responding to its request.
   After receiving a 100 response, the requestor SHOULD continue waiting
   for further responses. If more than 10 seconds elapses without
   receiving any response, the requestor MAY assume the responder is
   unresponsive and abort the connection.

   A requestor SHOULD wait longer than 10 seconds for a response if it
   is experiencing significant transport delays on its connection to the
   responder. The requestor is capable of determining the RTT using the
   Timestamp header (section 14.42) in any RTSP request.

9.6 Use of IPv6

   This specification has been updated so that it supports IPv6.
   However this support was not present in RFC 2326 therefore some
   interoperability issues exist. A RFC 2326 implementation can support



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   IPv6 as long as no explicit IPv6 addresses are used within RTSP
   messages. This require that any RTSP URL pointing at a IPv6 host must
   use fully qualified domain name and not a IPv6 address.  Further the
   Transport header must not use the parameters source and destination.

   Implementations according to this specification MUST understand IPv6
   addresses in URLs, and headers. By this requirement the feature-tag
   "play.basic" can be used to determine that a server or client is
   capable of handling IPv6 within RTSP.

10 Capability Handling

   This chapter describes the capability handling mechanism available in
   RTSP which allows RTSP to be extended. Extensions to this version of
   the protocol are basically done in two ways. First, new headers can
   be added. Secondly, new methods can be added. The capability handling
   mechanism is designed to handle these two cases.

   When a method is added the involved parties can use the OPTIONS
   method to discover if it is supported. This is done by issuing a
   OPTIONS request to the other party. Depending on the URL it will
   either apply in regards to a certain media resource, the whole server
   in general, or simply the next hop. The OPTIONS response will contain
   a Public header which declares all methods supported for the
   indicated resource.

   It is not necessary to use OPTIONS to discover support of a method,
   the client could simply try the method. If the receiver of the
   request does not support the method it will respond with an error
   code indicating the the method is either not implemented (501) or
   does not apply for the resource (405). The choice between the two
   discovery methods depends on the requirements of the service.

   To handle functionality additions that are not new methods feature-
   tags are defined. Each feature-tag represents a certain block of
   functionality. The amount of functionality that a feature-tag
   represents can vary significantly. A simple feature-tag can simple
   represent the functionality a single header gives. Another feature-
   tag is "play.basic" which represents the minimal playback
   implementation according to the updated specification.

   The feature-tags are then used to determine if the client, server or
   proxy supports the functionality that is necessary to achieve the
   desired service. To determine support of a feature-tag several
   different headers can be used, each explained below:

        Supported: The supported header is used to determine the
             complete set of functionality that both client and server



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             has. The intended usage is to determine before one needs to
             use a functionality that it is supported. It can be used in
             any method however OPTIONS is the most suitable as one at
             the same time determines all methods that are implemented.
             When sending a request the requestor declares all its
             capabilities by including all supported feature-tags. This
             results in that the receiver learns the requestors feature
             support. The receiver then includes its set of features in
             the response.

        Require: The Require header can be included in any request where
             the end point, i.e. the client or server, is required to
             understand the feature to correctly perform the request.
             This can for example be a SETUP request where the server
             must understand a certain parameter to be able to set up
             the media delivery correctly. Ignoring this parameter would
             not have the desired effect and is not acceptable.
             Therefore the end-point receiving a request containing a
             Require must negatively acknowledge any feature that it
             does not understand and not perform the request. The
             response in cases where features are not understood are 551
             (Option Not Supported). Also the features that are not
             understood are given in the Unsupported header in the
             response.

        Proxy-Require: This method has the same purpose and workings as
             Require except that it only applies to proxies and not the
             end point. Features that needs to be supported by both
             proxies and end-point needs to be included in both the
             Require and Proxy-Require header.

        Unsupported: This header is used in 551 error response to tell
             which feature(s) that was not supported. Such a response is
             only the result of the usage of the Require and/or Proxy-
             Require header where one or more feature where not
             supported. This information allows the requestor to make
             the best of situations as it knows which features that was
             not supported.

11 Method Definitions

   The method indicates what is to be performed on the resource
   identified by the Request-URL. The method name is case-sensitive.
   New methods may be defined in the future. Method names may not start
   with a $ character (decimal 24) and must be a token as defined by the
   ABNF. Methods are summarized in Table 11.





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    method         direction       object  Server req.    Client req.
    ___________________________________________________________________
    DESCRIBE       C -> S          P,S     recommended    recommended
    GET_PARAMETER  C -> S, S -> C  P,S     optional       optional
    OPTIONS        C -> S, S -> C  P,S     R=Req, Sd=Opt  Sd=Req, R=Opt
    PAUSE          C -> S          P,S     recommended    recommended
    PING           C -> S, S -> C  P,S     recommended    optional
    PLAY           C -> S          P,S     required       required
    REDIRECT       S -> C          P,S     optional       optional
    SETUP          C -> S          S       required       required
    SET_PARAMETER  C -> S, S -> C  P,S     optional       optional
    TEARDOWN       C -> S          P,S     required       required


   Table 7: Overview of RTSP methods, their direction, and what  objects  |
   (P:  presentation,  S:  stream)  they operate on. Legend:  R=Respond,  |
   Sd=Send, Opt: Optional, Req: Required, Rec:  Recommended

   Notes on Table 11: PAUSE is recommended, but not required. For         |
   example, a fully functional server can be built to deliver live feeds  |
   that does not support this method. If a RTSP agent does not support a  |
   particular method, it MUST return 501 (Not Implemented) and the        |
   requesting RTSP agent, in turn, SHOULD NOT try this method again for   |
   the given agent / resource combination.

11.1 OPTIONS

   The semantics of the RTSP OPTIONS method is equivalent to that of the  |
   HTTP OPTIONS method described in [H9.2]. In, RTSP, however, OPTIONS    |
   is bi-directional, where a client can request it to a server and vice  |
   versa. A client MUST implement the capability to send an OPTIONS       |
   request and a server or a proxy MUST implement the capability to       |
   respond to an OPTIONS request. The client, server or proxy MAY also    |
   implement the converse of their required capability.                   |

   An OPTIONS request may be issued at any time. Such a request does not  |
   modify the session state. However, it may prolong the session          |
   lifespan (see below). The URL in an OPTIONS request determines the     |
   scope of the request and the corresponding response.  If the request   |
   URL refers to a specific media resource on a given host, the scope is  |
   limited to the set of methods supported for that media resource by     |
   the indicated RTSP agent. A request URL with only the host address     |
   limits the scope to the specified RTSP agent's general capabilities    |
   without regard to any specific media. If the request URL is an         |
   asterisk ("*"), the scope is limited to the general capabilities of    |
   the next hop (i.e. the RTSP agent in direct communication with the     |
   request sender).                                                       |



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   Regardless of scope of the request, the Public header MUST always be   |
   included in the OPTIONS response listing the methods that are          |
   supported by the responding RTSP agent. In addition, if the scope of   |
   the request is limited to a media resource, the Allow header MAY be    |
   included in the response to enumerate the set of methods that are      |
   allowed for that resource. If the given resource is not available,     |
   the RTSP agent SHOULD return an appropriate response code such as 3rr  |
   or 4xx.  The Supported header can be included in the request to query  |
   the set of features that are supported by the responding RTSP agent.   |

   The OPTIONS method can be used to keep an RTSP session alive.          |
   However, it is not the preferred means of session keep-alive           |
   signalling, see section  14.40. An OPTIONS request intended for        |
   keeping alive a RTSP session MUST include the Session header with the  |
   associated session ID. Such a request SHOULD also use the media or     |
   the aggregated control URL as the request URL.

   Example:


     C->S:  OPTIONS * RTSP/1.0
            CSeq: 1
            User-Agent: PhonyClient/1.2
            Require:
            Proxy-Require: gzipped-messages
            Supported: play.basic

     S->C:  RTSP/1.0 200 OK
            CSeq: 1
            Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
            Supported: play.basic, implicit-play, gzipped-messages
            Server: PhonyServer/1.0



   Note that some of the feature-tags in Require and Proxy-Require are
   necessarily fictional features (one would hope that we would not
   purposefully overlook a truly useful feature just so that we could
   have a strong example in this section).

11.2 DESCRIBE

   The DESCRIBE method is used to retrieve the description of a           |
   presentation or media object from a server. The request URL of the     |
   DESCRIBE request identifies the media resource of interest. The        |
   client MAY include the Accept header in the request to list the        |
   description formats that it understands. The server SHALL respond      |
   with a description of the requested resource and return the            |



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   description in the entity of the response.  The DESCRIBE reply-
   response pair constitutes the media initialization phase of RTSP.

   Example:


     C->S: DESCRIBE rtsp://server.example.com/fizzle/foo RTSP/1.0
           CSeq: 312
           User-Agent: PhonyClient 1.2
           Accept: application/sdp, application/rtsl, application/mheg

     S->C: RTSP/1.0 200 OK
           CSeq: 312
           Date: 23 Jan 1997 15:35:06 GMT
           Server: PhonyServer 1.0
           Content-Type: application/sdp
           Content-Length: 376

           v=0
           o=mhandley 2890844526 2890842807 IN IP4 126.16.64.4
           s=SDP Seminar
           i=A Seminar on the session description protocol
           u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps
           e=mjh@isi.edu (Mark Handley)
           c=IN IP4 224.2.17.12/127
           t=2873397496 2873404696
           a=recvonly
           m=audio 3456 RTP/AVP 0
           m=video 2232 RTP/AVP 31
           m=application 32416 UDP WB
           a=orient:portrait



   The DESCRIBE response MUST contain all media initialization
   information for the resource(s) that it describes. Servers SHOULD NOT  |
   use the DESCRIBE response as a means of media indirection.             |


        By forcing a DESCRIBE response to contain all media          |
        initialization for the set of streams that it describes,     |
        and discouraging use of DESCRIBE for media indirection, we   |
        avoid looping problems that might result from other          |
        approaches.                                                  |

   Media initialization is a requirement for any RTSP-based system, but   |
   the RTSP specification does not dictate that this must be done via     |
   the DESCRIBE method. There are four ways that an RTSP client may       |



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   receive initialization information:                                    |


        o via a RTSP DESCRIBE method                                      |

        o via some other protocol (HTTP, email attachment, etc.)          |

        o via the command line or standard input                          |


   If a client obtains a valid description from an alternate source, the
   client MAY use this description for initialization purposes without
   issuing a DESCRIBE request for the same media.

   It is RECOMMENDED that minimal servers support the DESCRIBE method,
   and highly recommended that minimal clients support the ability to     |
   act as "helper applications" that accept a media initialization file   |
   from standard input, command line, and/or other means that are         |
   appropriate to the operating environment of the clients.

11.3 SETUP

   The SETUP request for a URL specifies the transport mechanism to be    |
   used for the streamed media. The SETUP method may be used in three     |
   different cases; Create a RTSP session, add a media to a session, and  |
   change the transport parameters of already set up media stream. Using  |
   SETUP to create or add media to a session when in PLAY state is        |
   unspecified. Otherwise SETUP can be used in all three states; INIT,    |
   and READY, for both purposes and in PLAY to change the transport       |
   parameters.

   The Transport header, see section  14.43, specifies the transport
   parameters acceptable to the client for data transmission; the
   response will contain the transport parameters selected by the
   server. This allows the client to enumerate in priority order the
   transport mechanisms and parameters acceptable to it, while the
   server can select the most appropriate. It is expected that the
   session description format used will enable the client to select a
   limited number possible configurations that are offered to the server
   to choose from. All transport parameters SHOULD be included in the
   Transport header, the use of other headers for this purpose is
   discouraged due to middle boxes.

   For the benefit of any intervening firewalls, a client SHOULD
   indicate the transport parameters even if it has no influence over
   these parameters, for example, where the server advertises a fixed
   multicast address.




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        Since SETUP includes all transport initialization
        information, firewalls and other intermediate network
        devices (which need this information) are spared the more
        arduous task of parsing the DESCRIBE response, which has
        been reserved for media initialization.

   In a SETUP response the server SHOULD include the Accept-Ranges
   header (see section 14.5 to indicate which time formats that are
   acceptable to use for this media resource.


     C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.0
           CSeq: 302
           Transport: RTP/AVP;unicast;client_port=4588-4589,
                      RTP/AVP/TCP;unicast;interleaved=0-1

     S->C: RTSP/1.0 200 OK
           CSeq: 302
           Date: 23 Jan 1997 15:35:06 GMT
           Server: PhonyServer 1.0
           Session: 47112344;timeout=60
           Transport: RTP/AVP;unicast;client_port=4588-4589;
                      server_port=6256-6257;ssrc=2A3F93ED
           Accept-Ranges: NPT



   In the above example the client want to create a RTSP session
   containing the media resource "rtsp://example.com/foo/bar/baz.rm".
   The transport parameters acceptable to the client is either
   RTP/AVP/UDP (UDP per default) to be received on client port 4588 and
   4589 or RTP/AVP interleaved on the RTSP control channel. The server
   selects the RTP/AVP/UDP transport and adds the ports it will send and
   received RTP and RTCP from, and the RTP SSRC that will be used by the
   server.

   The server MUST generate a session identifier in response to a
   successful SETUP request, unless a SETUP request to a server includes
   a session identifier, in which case the server MUST bundle this setup
   request into the existing session (aggregated session) or return
   error 459 (Aggregate Operation Not Allowed) (see Section  13.4.11).
   An Aggregate control URL MUST be used to control an aggregated
   session. This URL MUST be different from the stream control URLs of
   the individual media streams included in the aggregate. The Aggregate
   control URL is to be specified by the session description if the
   server supports aggregated control and aggregated control is desired
   for the session. However even if aggregated control is offered the
   client MAY chose to not set up the session in aggregated control. If



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   an Aggregate control URL is not specified in the session description,
   it is normally an indication that non-aggregated control should be
   used. The SETUP of media streams in an aggregate which has not been
   given an aggregated control URL is unspecified.


        While the session ID sometimes has enough information for
        aggregate control of a session, the Aggregate control URL
        is still important for some methods such as SET_PARAMETER
        where the control URL enables the resource in question to
        be easily identified. The Aggregate control URL is also
        useful for proxies, enabling them to route the request to
        the appropriate server, and for logging, where it is useful
        to note the actual resource that a request was operating
        on. Finally, presence of the Aggregate control URL allows
        for backwards compatibility with RFC 2326 [1].

   A session will exist until it is either removed by a TEARDOWN request
   or is timed-out by the server. The server MAY remove a session that
   has not demonstrated liveness signs from the client within a certain
   timeout period. The default timeout value is 60 seconds; the server
   MAY set this to a different value and indicate so in the timeout
   field of the Session header in the SETUP response. For further
   discussion see chapter  14.40. Signs of liveness for a RTSP session
   are:

        o Any RTSP request from a client which includes a Session header
          with that session's ID.

        o If RTP is used as a transport for the underlying media
          streams, an RTCP sender or receiver report from the client for
          any of the media streams in that RTSP session.

   If a SETUP request on a session fails for any reason, the session
   state, as well as transport and other parameters for associated
   streams SHALL remain unchanged from their values as if the SETUP
   request had never been received by the server.

   A client MAY issue a SETUP request for a stream that is already set
   up or playing in the session to change transport parameters, which a
   server MAY allow. If it does not allow this, it MUST respond with
   error 455 (Method Not Valid In This State). Reasons to support
   changing transport parameters, is to allow for application layer
   mobility and flexibility to utilize the best available transport as
   it becomes available.

   In a SETUP response for a request to change the transport parameters
   while in Play state, the server SHOULD include the Range to indicate



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   from what point the new transport parameters are used. Further if RTP
   is used for delivery the server SHOULD also include the RTP-Info
   header to indicate from what timestamp and RTP sequence number the
   change has taken place. If both RTP-Info and Range is included in the
   response the "rtp_time" parameter and range MUST be for the
   corresponding time, i.e. be used in the same way as for PLAY to
   ensure the correct synchronization information is present.

   If the transport parameter change while in PLAY state results in a
   change of synchronization related information, for example changing
   RTP SSRC, the server MUST provide in the SETUP response the necessary
   synchronization information. However the server is RECOMMENDED to
   avoid changing the synchronization information if possible.

11.4 PLAY

   The PLAY method tells the server to start sending data via the
   mechanism specified in SETUP. A client MUST NOT issue a PLAY request
   until any outstanding SETUP requests have been acknowledged as
   successful. PLAY requests are valid when the session is in READY
   state; the use of PLAY requests when the session is in PLAY state is
   deprecated. A PLAY request MUST include a Session header to indicate
   which session the request applies to.

   In an aggregated session the PLAY request MUST contain an aggregated
   control URL. A server SHALL responde with error 460 (Only Aggregate
   Operation Allowed) if the client PLAY request URL is for one of the
   media. The media in an aggregate SHALL be played in sync. If a client
   want individual control of the media it must use separate RTSP
   sessions for each media.

   The PLAY request SHALL position the normal play time to the beginning
   of the range specified by the Range header and delivers stream data
   until the end of the range if given, else to the end of the media is
   reached. To allow for precise composition multiple ranges MAY be
   specified in one PLAY Request. The range values are valid if all
   given ranges are part of any media within the aggregate. If a given
   range value points outside of the media, the response SHALL be the
   457 (Invalid Range) error code.

   The below example will first play seconds 10 through 15, then,
   immediately following, seconds 20 to 25, and finally seconds 30
   through the end.


     C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
           CSeq: 835
           Session: 12345678



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           Range: npt=10-15, npt=20-25, npt=30-



   See the description of the PAUSE request for further examples.

   A PLAY request without a Range header is legal. It SHALL start
   playing a stream from the beginning (npt=0-) unless the stream has
   been paused. If a stream has been paused via PAUSE, stream delivery
   resumes at the pause point. The stream SHALL play until the end of
   the media.

   The Range header MUST NOT contain a time parameter. The usage of time
   in PLAY method has been deprecated. If a request with time parameter
   is received the server SHOULD respond with a 457 (Invalid Range) to
   indicate that the time parameter is not supported.

   Server MUST include a "Range" header in any PLAY response. The
   response MUST use the same format as the request's range header
   contained. If no Range header was in the request, the NPT time format
   SHOULD be used unless the client showed support for an other format
   more appropriate. Also for a session with live media streams the
   Range header MUST indicate a valid time. It is RECOMMENDED that
   normal play time is used, either the "now" indicator, for example
   "npt=now-", or the time since session start as an open interval, e.g.
   "npt=96.23-". An absolute time value (clock) for the corresponding
   time MAY be given, i.e.  "clock=20030213T143205Z-". The UTC clock
   format SHOULD only be used if client has shown support for it.

   A media server only supporting playback MUST support the npt format
   and MAY support the clock and smpte formats.

   For a on-demand stream, the server MUST reply with the actual range    |
   that will be played back, i.e. for which duration all media having     |
   content at this time is delivered. This may differ from the requested  |
   range if alignment of the requested range to valid frame boundaries    |
   is required for the media source. Note that some media streams in an   |
   aggregate may need to be delivered from even earlier points. Also      |
   some media format has very long duration per individual data unit,     |
   therefore it might be necessary for the client to parse the data       |
   unit, and select where to start.                                       |


   Example: Single audio stream (MIDI)                                    |

   C->S: PLAY rtsp://example.com/audio RTSP/1.0                           |
         CSeq: 836                                                        |
         Session: 12345678                                                |



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         Range: npt=7.05-                                                 |

   S->C: RTSP/1.0 200 OK                                                  |
         CSeq: 836                                                        |
         Date: 23 Jan 1997 15:35:06 GMT                                   |
         Server: PhonyServer 1.0                                          |
         Range: npt=3.52-                                                 |
         RTP-Info:url=rtsp://example.com/audio;                           |
            seq=14783;rtptime=2345962545                                  |

   S->C: RTP Packet TS=2345962545 => NPT=3.52                             |
         Duration: 4.15 seconds                                           |



   In this example the client receives the first media packet that        |
   stretches all the way up and past the requested playtime. Thus it is   |
   a client decision if it desires to render to the user the time         |
   between 3.52 and 7.05 that the user requested. In most cases it is     |
   probably suitable to not render that time period.                      |

   For live media sources it might be impossible to specify from which    |
   point in time all media streams that has active content can actually   |
   be delivered. Therefore a server MAY specify a start time (or now-)    |
   in the range header, for which not all media will be available from.

   If no range is specified in the request, the start position SHALL
   still be returned in the reply. If the medias that are part of an
   aggregate has different lengths, the PLAY request SHALL be performed
   as long as the given range is valid for any media, for example the
   longest media. Media will be sent whenever it is available for the
   given play-out point.

   A PLAY response MAY include a header(s) carrying synchronization       |
   information. As the information necessary is dependent on the media    |
   transport format, further rules specifying the header and its usage    |
   is needed. For RTP the RTP-Info header is specified, see section       |
   14.36.

   After playing the desired range, the presentation is NOT
   automatically paused, media delivery simply stops. A PAUSE request
   MUST be issued before another PLAY request can be issued. Note: This
   is one change resulting in a non-operability with RFC 2326
   implementations. A client not issuing a PAUSE request before a new
   PLAY will be stuck in PLAY state. To mitigate this backwards
   compatibility issue the following behavior is recommended.  If a
   server receives a PLAY request when in play state and all media has
   finished the requested play out, the server MAY interpret this as a



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   PLAY request received in ready state. However the server SHALL NOT do
   this if the client has shown any support for this specification, for
   example by sending a Supported header with the play.basic feature
   tag.

   A client desiring to play the media from the beginning MUST send a
   PLAY request with a Range header pointing at the beginning, e.g.
   npt=0-. If a PLAY request is received without a Range header when
   media delivery has stopped at the end, the server SHOULD respond with
   a 457 "Invalid Range" error response. In that response the current
   pause point in a Range header SHALL be included.

   The following example plays the whole presentation starting at SMPTE
   time code 0:10:20 until the end of the clip. Note: The RTP-Info
   headers has been broken into several lines to fit the page.


   C->S: PLAY rtsp://audio.example.com/twister.en RTSP/1.0
         CSeq: 833
         Session: 12345678
         Range: smpte=0:10:20-

   S->C: RTSP/1.0 200 OK
         CSeq: 833
         Date: 23 Jan 1997 15:35:06 GMT
         Server: PhonyServer 1.0
         Range: smpte=0:10:22-0:15:45
         RTP-Info:url=rtsp://example.com/twister.en;
            seq=14783;rtptime=2345962545



   For playing back a recording of a live presentation, it may be
   desirable to use clock units:


     C->S: PLAY rtsp://audio.example.com/meeting.en RTSP/1.0
           CSeq: 835
           Session: 12345678
           Range: clock=19961108T142300Z-19961108T143520Z

     S->C: RTSP/1.0 200 OK
           CSeq: 835
           Date: 23 Jan 1997 15:35:06 GMT
           Server:PhonyServer 1.0
           Range: clock=19961108T142300Z-19961108T143520Z
           RTP-Info:url=rtsp://example.com/meeting.en;
              seq=53745;rtptime=484589019



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   All range specifiers in this specification allow for ranges with
   unspecified begin times (e.g. "npt=-30"). When used in a PLAY
   request, the server treats this as a request to start/resume playback
   from the current pause point, ending at the end time specified in the
   Range header. If the pause point is located later than the given end
   value, a 457 (Invalid Range) response SHALL be given.

   The queued play functionality described in RFC 2326 [1] is removed
   and multiple ranges can be used to achieve a similar functionality.
   If a server receives a PLAY request while in the PLAY state, the
   server SHALL responde using the error code 455 (Method Not Valid In
   This State). This will signal the client that queued play are not
   supported.

   The use of PLAY for keep-alive signaling, i.e. PLAY request without a
   range header in PLAY state, has also been depreciated. Instead a
   client can use, PING, SET_PARAMETER or OPTIONS for keep alive. A
   server receiving a PLAY keep alive SHALL respond with the 455 error
   code.

11.5 PAUSE

   The PAUSE request causes the stream delivery to be interrupted
   (halted) temporarily. A PAUSE request MUST be done with the
   aggregated control URL for aggregated sessions, resulting in all
   media being halted, or the media URL for non-aggregated sessions.
   Any attempt to do muting of a single media with an PAUSE request in
   an aggregated session SHALL be responded with error 460 (Only
   Aggregate Operation Allowed). After resuming playback,
   synchronization of the tracks MUST be maintained. Any server
   resources are kept, though servers MAY close the session and free
   resources after being paused for the duration specified with the
   timeout parameter of the Session header in the SETUP message.

   Example:


     C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 834
           Session: 12345678

     S->C: RTSP/1.0 200 OK
           CSeq: 834
           Date: 23 Jan 1997 15:35:06 GMT
           Range: npt=45.76-






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   The PAUSE request MAY contain a Range header specifying when the
   stream or presentation is to be halted. We refer to this point as the
   "pause point". The time parameter in the Range MUST NOT be used. The
   Range header MUST contain a single value, expressed as the beginning
   value an open range. For example, the following clip will be played
   from 10 seconds through 21 seconds of the clip's normal play time,
   under the assumption that the PAUSE request reaches the server within
   11 seconds of the PLAY request. Note that some lines has been broken
   in an non-correct way to fit the page:


     C->S: PLAY rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 834
           Session: 12345678
           Range: npt=10-30

     S->C: RTSP/1.0 200 OK
           CSeq: 834
           Date: 23 Jan 1997 15:35:06 GMT
           Server: PhonyServer 1.0
           Range: npt=10-30
           RTP-Info:url=rtsp://example.com/fizzle/audiotrack;
                   seq=5712;rtptime=934207921,
                   url=rtsp://example.com/fizzle/videotrack;
                   seq=57654;rtptime=2792482193
           Session: 12345678

     C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 835
           Session: 12345678
           Range: npt=21-

     S->C: RTSP/1.0 200 OK
           CSeq: 835
           Date: 23 Jan 1997 15:35:09 GMT
           Server: PhonyServer 1.0
           Range: npt=21-
           Session: 12345678



   The pause request becomes effective the first time the server is
   encountering the time point specified in any of the multiple ranges.
   If the Range header specifies a time outside any range from the PLAY
   request, the error 457 (Invalid Range) SHALL be returned. If a media
   unit (such as an audio or video frame) starts presentation at exactly
   the pause point, it is not played. If the Range header is missing,
   stream delivery is interrupted immediately on receipt of the message



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   and the pause point is set to the current normal play time. However,
   the pause point in the media stream MUST be maintained. A subsequent
   PLAY request without Range header SHALL resume from the pause point
   and play until media end.

   If the server has already sent data beyond the time specified in the
   PAUSE request's Range header, a PLAY without range SHALL resume at
   the point in time specified by the PAUSE request's Range header, as
   it is assumed that the client has discarded data after that point.
   This ensures continuous pause/play cycling without gaps.

   The pause point after any PAUSE request SHALL be returned to the
   client by adding a Range header with what remains unplayed of the
   PLAY request's ranges, i.e. including all the remaining ranges part
   of multiple range specification. If one desires to resume playing a
   ranged request, one simply includes the Range header from the PAUSE
   response. Note that this server behavior was not mandated previously
   and servers implementing according to RFC 2326 will probably not
   return the range header.

   For example, if the server have a play request for ranges 10 to 15
   and 20 to 29 pending and then receives a pause request for NPT 21, it
   would start playing the second range and stop at NPT 21. If the pause
   request is for NPT 12 and the server is playing at NPT 13 serving the
   first play request, the server stops immediately. If the pause
   request is for NPT 16, the server returns a 457 error message. To
   prevent that the second range is played and the server stops after
   completing the first range, a PAUSE request for 20 must be issued.

   As another example, if a server has received requests to play ranges
   10 to 15 and then 13 to 20 (that is, overlapping ranges), the PAUSE
   request for NPT=14 would take effect while the server plays the first
   range, with the second range effectively being ignored, assuming the
   PAUSE request arrives before the server has started playing the
   second, overlapping range. Regardless of when the PAUSE request
   arrives, it sets the pause point to 14. The below example messages is
   for the above case when the PAUSE request arrives before the first
   occurrence of NPT=14.


     C->S: PLAY rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 834
           Session: 12345678
           Range: npt=10-15, npt=13-20

     S->C: RTSP/1.0 200 OK
           CSeq: 834
           Date: 23 Jan 1997 15:35:06 GMT



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           Server: PhonyServer 1.0
           Range: npt=10-15, npt=13-20
           RTP-Info:url=rtsp://example.com/fizzle/audiotrack;
                   seq=5712;rtptime=934207921,
                   url=rtsp://example.com/fizzle/videotrack;
                   seq=57654;rtptime=2792482193
           Session: 12345678

     C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 835
           Session: 12345678
           Range: npt=14-

     S->C: RTSP/1.0 200 OK
           CSeq: 835
           Date: 23 Jan 1997 15:35:09 GMT
           Server: PhonyServer 1.0
           Range: npt=14-15, npt=13-20
           Session: 12345678



   If a client issues a PAUSE request and the server acknowledges and
   enters the READY state, the proper server response, if the player
   issues another PAUSE, is still 200 OK. The 200 OK response MUST
   include the Range header with the current pause point, even if the
   PAUSE request is asking for some other pause point. See examples
   below:

   Examples:

     C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 834
           Session: 12345678

     S->C: RTSP/1.0 200 OK
           CSeq: 834
           Session: 12345678
           Date: 23 Jan 1997 15:35:06 GMT
           Range: npt=45.76-98.36

     C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 835
           Session: 12345678
           Range: 86-

     S->C: RTSP/1.0 200 OK
           CSeq: 835



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           Session: 12345678
           Date: 23 Jan 1997 15:35:07 GMT
           Range: npt=45.76-98.36



11.6 TEARDOWN

   The TEARDOWN client to server request stops the stream delivery for
   the given URL, freeing the resources associated with it.  TEARDOWN
   MAY be done using either an aggregated or a media control URL.
   However some restrictions apply depending on the current state. The
   TEARDOWN request SHALL contain a Session header indicating what
   session the request applies to.

   A TEARDOWN using the aggregated control URL or the media URL in a
   session under non-aggregated control MAY be done in any state (Ready,
   and Play). A successful request SHALL result in that media delivery
   is immediately halted and the session state is destroyed. This SHALL
   be indicated through the lack of a Session header in the response.

   A TEARDOWN using a media URL in an aggregated session MAY only be      |
   done in Ready state. Such a request only removes the indicated media   |
   stream and associated resources from the session.  This may result in  |
   that a session returns to non-aggregated control, due to that it only  |
   contains a single media after the requests completion. In the          |
   response to TEARDOWN request resulting in that the session still       |
   exist SHALL contain a Session header to indicate this.

   Note, the indication with the session header if sessions state remain
   may not be done correctly by a RFC 2326 client, but will be for any
   server signalling the "play.basic" tag.

   Example:


     C->S: TEARDOWN rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 892
           Session: 12345678

     S->C: RTSP/1.0 200 OK
           CSeq: 892
           Server: PhonyServer 1.0



11.7 GET_PARAMETER




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   The GET_PARAMETER request retrieves the value of a parameter or
   parameters for a presentation or stream specified in the URL. If the
   Session header is present in a request, the value of a parameter MUST
   be retrieved in the specified session context.  The content of the
   reply and response is left to the implementation.

   The method MAY also be used without a body (entity). If the this
   request is successful, i.e. a 200 OK response is received, then the
   keep-alive time has been updated. Any non-required header present in
   such a request, may or may not been processed. The allow a client to
   determine if any such header has been processed, it is necessary to
   use a feature tag and the Require header. Due to this reason it is
   RECOMMENDED that any parameters to be retrieved are sent in the body,
   rather than using any header.

   Example:


     S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 431
           Content-Type: text/parameters
           Session: 12345678
           Content-Length: 15

           packets_received
           jitter

     C->S: RTSP/1.0 200 OK
           CSeq: 431
           Content-Length: 46
           Content-Type: text/parameters

           packets_received: 10
           jitter: 0.3838




        The "text/parameters" section is only an example type for
        parameter. This method is intentionally loosely defined
        with the intention that the reply content and response
        content will be defined after further experimentation.

11.8 SET_PARAMETER

   This method requests to set the value of a parameter or a set of
   parameters for a presentation or stream specified by the URL. The
   method MAY also be used without a body (entity). If the this request



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   is successful, i.e. a 200 OK response is received, then the keep-
   alive time has been updated. Any non-required header present in such
   a request, may or may not been processed. The allow a client to
   determine if any such header has been processed, it is necessary to
   use a feature tag and the Require header. Due to this reason it is
   RECOMMENDED that any parameters are sent in the body, rather than
   using any header.

   A request is RECOMMENDED to only contain a single parameter to allow
   the client to determine why a particular request failed. If the
   request contains several parameters, the server MUST only act on the
   request if all of the parameters can be set successfully. A server
   MUST allow a parameter to be set repeatedly to the same value, but it
   MAY disallow changing parameter values.  If the receiver of the
   request does not understand or can locate a parameter error 451
   (Parameter Not Understood) SHALL be used.  In the case a parameter is
   not allowed to change the error code 458 (Parameter Is Read-Only).
   The response body SHOULD contain only the parameters that has errors.
   Otherwise no body SHALL be returned.

   Note: transport parameters for the media stream MUST only be set with
   the SETUP command.

        Restricting setting transport parameters to SETUP is for
        the benefit of firewalls.


        The parameters are split in a fine-grained fashion so that
        there can be more meaningful error indications. However, it
        may make sense to allow the setting of several parameters
        if an atomic setting is desirable. Imagine device control
        where the client does not want the camera to pan unless it
        can also tilt to the right angle at the same time.

   Example:


     C->S: SET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 421
           Content-length: 20
           Content-type: text/parameters

           barparam: barstuff

     S->C: RTSP/1.0 451 Parameter Not Understood
           CSeq: 421
           Content-length: 10
           Content-type: text/parameters



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           barparam




        The "text/parameters" section is only an example type for
        parameter. This method is intentionally loosely defined
        with the intention that the reply content and response
        content will be defined after further experimentation.

11.9 REDIRECT


   The REDIRECT method is issued by a server to inform a client that it   |
   MUST connect to another server location to access the resource         |
   indicated by the Request URL. The presence of the Session header in a  |
   REDIRECT request indicates the scope of the request, and determines    |
   the specific semantics of the request.                                 |

   A REDIRECT request with a Session header has end-to-end (i.e. server   |
   to client) scope and applies only to the given session.  Any           |
   intervening proxies SHOULD NOT disconnect the control channel while    |
   there are other remaining end-to-end sessions. The OPTIONAL Location   |
   header, if included in such a request, SHALL contain a complete        |
   absolute URL pointing to the resource to which the client SHOULD       |
   reconnect. Specifically, the Location SHALL NOT contain just the host  |
   and port. A client may receive a REDIRECT request with a Session       |
   header, if and only if, an end-to-end session has been established.    |

   A client may receive a REDIRECT request without a Session header at    |
   any time when it has communication or a connection established with a  |
   server. The scope of such a request is limited to the next-hop (i.e.   |
   the RTSP agent in direct communication with the server) and applies,   |
   as well, to the control connection between the next-hop RTSP agent     |
   and the server. A REDIRECT request without a Session header indicates  |
   that all sessions and pending requests being managed via the control   |
   connection MUST be redirected. The OPTIONAL Location header, if        |
   included in such a request, SHOULD contain an absolute URL with only   |
   the host address and the OPTIONAL port number of the server to which   |
   the RTSP agent SHOULD reconnect. Any intervening proxies SHOULD do     |
   all of the following in the order listed:                              |

        1.   respond to the REDIRECT request                              |

        2.   disconnect the control channel from the requesting server    |

        3.   connect to the server at the given host address              |




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        4.   pass the REDIRECT request to each applicable client          |
             (typically those clients with an active session or an        |
             unanswered request)                                          |

   Note: The proxy is responsible for accepting REDIRECT responses from   |
   its clients; these responses MUST NOT be passed on to either the       |
   original server or the redirected server.                              |

   The lack of a Location header in any REDIRECT request is indicative    |
   of the server no longer being able to fulfill the current request and  |
   having no alternatives for the client to continue with its normal      |
   operation. It is akin to a server initiated TEARDOWN that applies      |
   both to sessions as well as the general connection associated with     |
   that client.                                                           |

   When the Range header is not included in a REDIRECT request, the       |
   client SHOULD perform the redirection immediately and return a         |
   response to the server. The server can consider the session as         |
   terminated and can free any associated state after it receives the     |
   successful (2xx) response. The server MAY close the signalling         |
   connection upon receiving the response and the client SHOULD close     |
   the signalling connection after sending the 2xx response. The          |
   exception to this is when the client has several sessions on the       |
   server being managed by the given signalling connection. In this       |
   case, the client SHOULD close the connection when it has received and  |
   responded to REDIRECT requests for all the sessions managed by the     |
   signalling connection.                                                 |

   If the OPTIONAL Range header is included in a REDIRECT request, it     |
   indicates when the redirection shall take effect. The range value      |
   MUST be an open ended single value, e.g.  npt=59-, indicating the      |
   play out time when redirection SHALL occur. Alternatively, a range     |
   with a time= parameter indicates the wall clock time by when the       |
   redirection MUST take place. When the time= parameter is present in    |
   the range, any range value MUST be ignored even though it MUST be      |
   syntactically correct. When the indicated redirect point is reached,   |
   a client MUST issue a TEARDOWN request and SHOULD close the            |
   signalling connection after receiving a 2xx response. The normal       |
   connection considerations apply for the server.                        |


        The differentiation of REDIRECT requests with and without    |
        range headers is to allow for clear and explicit state       |
        handling. As the state in the server needs to be kept until  |
        the point of redirection, the handling becomes more clear    |
        if the client is required to TEARDOWN the session at the     |
        redirect point.                                              |




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   After a REDIRECT request has been processed, a client that wants to    |
   continue to send or receive media for the resource identified by the   |
   request URL will have to establish a new session with the designated   |
   host. If the URL given in the Location header is a valid resource      |
   URL, a client SHOULD issue a DESCRIBE request for the URL.             |

        Note: The media resource indicated by the Location header    |
        can be either identical, slightly different or totally       |
        different. This is the reason why a new DESCRIBE request     |
        SHOULD be issued.  If the Location header contains only a    |
        host address, the client MAY assume that the media on the    |
        new server is identical to the media on the old server,      |
        i.e. all media configuration information from the old        |
        session is still valid except for the host address.

   This example request redirects traffic for this session to the new
   server at the given absolute time:


     S->C: REDIRECT rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 732
           Location: rtsp://s2.example.com:8001
           Range: npt=0- ;time=19960213T143205Z
           Session: uZ3ci0K+Ld-M



11.10 PING

   This method is a bi-directional mechanism for server or client
   liveness checking. It has no side effects. The issuer of the request
   MUST include a session header with the session ID of the session that
   is being checked for liveness.

   Prior to using this method, an OPTIONS method is RECOMMENDED to be
   issued in the direction which the PING method would be used. This
   method MUST NOT be used if support is not indicated by the Public
   header. Note: That an 501 (Not Implemented) response means that the
   keep-alive timer has not been updated.

   When a proxy is in use, PING with a * indicates a single-hop liveness
   check, whereas PING with a URL including an host address indicates an
   end-to-end liveness check.

   Example:

     C->S: PING * RTSP/1.0
           CSeq: 123



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           Session:12345678

     S->C: RTSP/1.0 200 OK
           CSeq: 123
           Session:12345678



12 Embedded (Interleaved) Binary Data

   Certain firewall designs and other circumstances may force a server
   to interleave RTSP messages and media stream data. This interleaving
   should generally be avoided unless necessary since it complicates
   client and server operation and imposes additional overhead. Also
   head of line blocking may cause problems.  Interleaved binary data
   SHOULD only be used if RTSP is carried over TCP.

   Stream data such as RTP packets is encapsulated by an ASCII dollar
   sign (24 decimal), followed by a one-byte channel identifier,
   followed by the length of the encapsulated binary data as a binary,
   two-byte integer in network byte order. The stream data follows
   immediately afterwards, without a CRLF, but including the upper-layer
   protocol headers. Each $ block SHALL contain exactly one upper-layer
   protocol data unit, e.g., one RTP packet.




       0                   1                   2                   3
       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      | "$" = 24      | Channel ID    | Length in bytes               |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      : Length number of bytes of binary data                         :
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+





   The channel identifier is defined in the Transport header with the
   interleaved parameter(Section 14.43).

   When the transport choice is RTP, RTCP messages are also interleaved
   by the server over the TCP connection. The usage of RTCP messages is
   indicated by including a range containing a second channel in the
   interleaved parameter of the Transport header, see section 14.43. If
   RTCP is used, packets SHALL be sent on the first available channel



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   higher than the RTP channel. The channels are bi-directional and
   therefore RTCP traffic are sent on the second channel in both
   directions.


        RTCP is needed for synchronization when two or more streams
        are interleaved in such a fashion. Also, this provides a
        convenient way to tunnel RTP/RTCP packets through the TCP
        control connection when required by the network
        configuration and transfer them onto UDP when possible.



     C->S: SETUP rtsp://example.com/bar.file RTSP/1.0
           CSeq: 2
           Transport: RTP/AVP/TCP;unicast;interleaved=0-1

     S->C: RTSP/1.0 200 OK
           CSeq: 2
           Date: 05 Jun 1997 18:57:18 GMT
           Transport: RTP/AVP/TCP;unicast;interleaved=5-6
           Session: 12345678

     C->S: PLAY rtsp://example.com/bar.file RTSP/1.0
           CSeq: 3
           Session: 12345678

     S->C: RTSP/1.0 200 OK
           CSeq: 3
           Session: 12345678
           Date: 05 Jun 1997 18:59:15 GMT
           RTP-Info: url=rtsp://example.com/bar.file;
             seq=232433;rtptime=972948234

     S->C: $005{2 byte length}{"length" bytes data, w/RTP header}
     S->C: $005{2 byte length}{"length" bytes data, w/RTP header}
     S->C: $006{2 byte length}{"length" bytes  RTCP packet}



13 Status Code Definitions

   Where applicable, HTTP status [H10] codes are reused. Status codes
   that have the same meaning are not repeated here. See Table 4 for a
   listing of which status codes may be returned by which requests. All
   error messages, 4xx and 5xx MAY return a body containing further
   information about the error.




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13.1 Success 1xx

13.1.1 100 Continue

   See, [H10.1.1].

13.2 Success 2xx

13.3 Redirection 3xx

   The notation "3rr" indicates response codes from 300 to 399 inclusive
   which are meant for redirection. The response code 304 is excluded
   from this set, as it is not used for redirection.

   See [H10.3] for definition of status code 300 to 305. However
   comments are given for some to how they apply to RTSP.

   Within RTSP, redirection may be used for load balancing or
   redirecting stream requests to a server topologically closer to the
   client.  Mechanisms to determine topological proximity are beyond the
   scope of this specification.

   A 3rr code MAY be used to respond to any request. It is RECOMMENDED
   that they are used if necessary before a session is established, i.e.
   in response to DESCRIBE or SETUP. However in cases where a server is
   not able to send a REDIRECT request to the client, the server MAY
   need to resort to using 3rr responses to inform a client with a
   established session about the need for redirecting the session. If an
   3rr response is received for an request in relation to a established
   session, the client SHOULD send a TEARDOWN request for the session,
   and MAY reestablish the session using the resource indicated by the
   Location.

   If the the Location header is used in a response it SHALL contain an
   absolute URL pointing out the media resource the client is redirected
   to, the URL SHALL NOT only contain the host name.

13.3.1 300 Multiple Choices

13.3.2 301 Moved Permanently

   The request resource are moved permanently and resides now at the URL
   given by the location header. The user client SHOULD redirect
   automatically to the given URL. This response MUST NOT contain a
   message-body.

13.3.3 302 Found




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   The requested resource reside temporarily at the URL given by the
   Location header. The Location header MUST be included in the
   response. Is intended to be used for many types of temporary
   redirects, e.g. load balancing. It is RECOMMENDED that one set the
   reason phrase to something more meaningful than "Found" in these
   cases. The user client SHOULD redirect automatically to the given
   URL. This response MUST NOT contain a message-body.

13.3.4 303 See Other

   This status code SHALL NOT be used in RTSP. However as it was allowed
   to use in RFC 2326 it is possible that such response may be received,
   in which case the behavior is undefined.

13.3.5 304 Not Modified

   If the client has performed a conditional DESCRIBE or SETUP (see
   12.23) and the requested resource has not been modified, the server
   SHOULD send a 304 response. This response MUST NOT contain a
   message-body.

   The response MUST include the following header fields:

        o Date

        o ETag and/or Content-Location, if the header would have been
          sent in a 200 response to the same request.

        o Expires, Cache-Control, and/or Vary, if the field-value might
          differ from that sent in any previous response for the same
          variant.

   This response is independent for the DESCRIBE and SETUP requests.
   That is, a 304 response to DESCRIBE does NOT imply that the resource
   content is unchanged and a 304 response to SETUP does NOT imply that
   the resource description is unchanged. The ETag and If-Match headers
   may be used to link the DESCRIBE and SETUP in this manner.

13.3.6 305 Use Proxy

   See [H10.3.6].

13.4 Client Error 4xx

13.4.1 400 Bad Request

   The request could not be understood by the server due to malformed
   syntax. The client SHOULD NOT repeat the request without



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   modifications [H10.4.1]. If the request does not have a CSeq header,
   the server MUST NOT include a CSeq in the response.

13.4.2 405 Method Not Allowed

   The method specified in the request is not allowed for the resource
   identified by the request URL. The response MUST include an Allow
   header containing a list of valid methods for the requested resource.
   This status code is also to be used if a request attempts to use a
   method not indicated during SETUP, e.g., if a RECORD request is
   issued even though the mode parameter in the Transport header only
   specified PLAY.

13.4.3 451 Parameter Not Understood

   The recipient of the request does not support one or more parameters
   contained in the request.When returning this error message the sender
   SHOULD return a entity body containing the offending parameter(s).

13.4.4 452 reserved

   This error code was removed from RFC 2326 [1] and is obsolete.

13.4.5 453 Not Enough Bandwidth

   The request was refused because there was insufficient bandwidth.
   This may, for example, be the result of a resource reservation
   failure.

13.4.6 454 Session Not Found

   The RTSP session identifier in the Session header is missing,
   invalid, or has timed out.

13.4.7 455 Method Not Valid in This State

   The client or server cannot process this request in its current
   state.  The response SHOULD contain an Allow header to make error
   recovery easier.

13.4.8 456 Header Field Not Valid for Resource

   The server could not act on a required request header. For example,
   if PLAY contains the Range header field but the stream does not allow
   seeking. This error message may also be used for specifying when the
   time format in Range is impossible for the resource. In that case the
   Accept-Ranges header SHOULD be returned to inform the client of which
   format(s) that are allowed.



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13.4.9 457 Invalid Range

   The Range value given is out of bounds, e.g., beyond the end of the
   presentation.

13.4.10 458 Parameter Is Read-Only

   The parameter to be set by SET_PARAMETER can be read but not
   modified. When returning this error message the sender SHOULD return
   a entity body containing the offending parameter(s).

13.4.11 459 Aggregate Operation Not Allowed

   The requested method may not be applied on the URL in question since
   it is an aggregate (presentation) URL. The method may be applied on a
   media URL.

13.4.12 460 Only Aggregate Operation Allowed

   The requested method may not be applied on the URL in question since
   it is not an aggregate control (presentation) URL. The method may be
   applied on the aggregate control URL.

13.4.13 461 Unsupported Transport

   The Transport field did not contain a supported transport
   specification.

13.4.14 462 Destination Unreachable

   The data transmission channel could not be established because the
   client address could not be reached. This error will most likely be
   the result of a client attempt to place an invalid Destination
   parameter in the Transport field.

13.4.15 470 Connection Authorization Required

   The secured connection attempt need user or client authorization
   before proceeding. The next hops certificate is included in this
   response in the Accept-Credentials header.

13.4.16 471 Connection Credentials not accepted

   When performing a secure connection over multiple connections, a
   intermediary has refused to connect to the next hop and carry out the
   request due to unacceptable credentials for the used policy.

13.5 Server Error 5xx



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13.5.1 551 Option not supported

   An feature-tag given in the Require or the Proxy-Require fields was
   not supported. The Unsupported header SHOULD be returned stating the
   feature for which there is no support.

14 Header Field Definitions


             method        direction      object acronym Body
             _________________________________________________
             DESCRIBE      C -> S         P,S    DES     r
             GET_PARAMETER C -> S, S -> C P,S    GPR     R,r
             OPTIONS       C -> S         P,S    OPT
                           S -> C
             PAUSE         C -> S         P,S    PSE
             PING          C -> S, S -> C P,S    PNG
             PLAY          C -> S         P,S    PLY
             REDIRECT      S -> C         P,S    RDR
             SETUP         C -> S         S      STP
             SET_PARAMETER C -> S, S -> C P,S    SPR     R,r
             TEARDOWN      C -> S         P,S    TRD


   Table 8: Overview of RTSP methods, their direction, and what  objects
   (P:  presentation, S: stream) they operate on. Body notes if a method
   is allowed to carry  body  and  in  which  direction,  R  =  Request,
   r=response. Note: It is allowed for all error messages 4xx and 5xx to
   have a body


   The general syntax for header fields is covered in Section 4.2 This
   section lists the full set of header fields along with notes on
   meaning, and usage. The syntax definition for headers are present in
   section 18.2.3. Throughout this section, we use [HX.Y] to refer to
   Section X.Y of the current HTTP/1.1 specification RFC 2616 [4].
   Examples of each header field are given.

   Information about header fields in relation to methods and proxy
   processing is summarized in Tables  9, 10,  11, and 12.

   The "where" column describes the request and response types in which
   the header field can be used. Values in this column are:

        R: header field may only appear in requests;

        r: header field may only appear in responses;




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        2xx, 4xx, etc.: A numerical value or range indicates response
             codes with which the header field can be used;

        c: header field is copied from the request to the response.

   An empty entry in the "where" column indicates that the header field
   may be present in all requests and responses.

   The "proxy" column describes the operations a proxy may perform on a
   header field:

        a: A proxy can add or concatenate the header field if not
             present.

        m: A proxy can modify an existing header field value.

        d: A proxy can delete a header field value.

        r: A proxy must be able to read the header field, and thus this
             header field cannot be encrypted.

   The rest of the columns relate to the presence of a header field in a
   method. The method names when abbreviated, are according to table 8:

        c: Conditional; requirements on the header field depend on the
             context of the message.

        m: The header field is mandatory.

        m*: The header field SHOULD be sent, but clients/servers need to
             be prepared to receive messages without that header field.

        o: The header field is optional.

        *: The header field is required if the message body is not
             empty. See sections 14.15, 14.17 and 4.3 for details.

        -: The header field is not applicable.

   "Optional" means that a Client/Server MAY include the header field in
   a request or response. The Client/Server behavior when receiving such
   headers varies, for some it may ignore the header field, in other
   case it is request to process the header. This is regulated by the
   method and header descriptions. Example of such headers that require
   processing are the Require and Proxy-Require header fields discussed
   in 14.35 and 14.30. A "mandatory" header field MUST be present in a
   request, and MUST be understood by the Client/Server receiving the
   request. A mandatory response header field MUST be present in the



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   response, and the header field MUST be understood by the
   Client/Server processing the response. "Not applicable" means that
   the header field MUST NOT be present in a request. If one is placed
   in a request by mistake, it MUST be ignored by the Client/Server
   receiving the request. Similarly, a header field labeled "not
   applicable" for a response means that the Client/Server MUST NOT
   place the header field in the response, and the Client/Server MUST
   ignore the header field in the response.

   A Client/Server SHOULD ignore extension header parameters that are
   not understood.

   The From, Location, and RTP-Info header fields contain a URL. If the
   URL contains a comma, or semicolon, the URL MUST be enclosed in
   double quotas ("). Any URL parameters are contained within these
   quotas. If the URL is not enclosed in double quotas, any semicolon-
   delimited parameters are header-parameters, not URL parameters.





14.1 Accept

   The Accept request-header field can be used to specify certain
   presentation description content types which are acceptable for the
   response.

        The "level" parameter for presentation descriptions is
        properly defined as part of the MIME type registration, not
        here.

   See [H14.1] for syntax.

   Example of use:


     Accept: application/rtsl q=1.0, application/sdp



14.2 Accept-Credentials


   The Accept-Credentials header has different usage in RTSP requests     |
   and responses. In a request it is used to indicate to any trusted      |
   intermediary how to handle further secured connections to proxies or   |
   servers. In responses the header is used to carry the credentials of   |



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   Header              Where  Proxy DES OPT SETUP PLAY PAUSE TRD
   _____________________________________________________________
   Accept                R           o   -    -    -     -   -
   Accept-Credentials    R      r    o   o    o    o     o   o
   Accept-Credentials 470,407  ar    o   o    o    o     o   o
   Accept-Encoding       R      r    o   -    -    -     -   -
   Accept-Language       R      r    o   -    -    -     -   -
   Accept-Ranges         r      r    -   -    o    -     -   -
   Accept-Ranges        456     r    -   -    -    o     o   -
   Allow                 r           -   o    -    -     -   -
   Allow                405          m   m    m    m     m   m
   Authorization         R           o   o    o    o     o   o
   Bandwidth             R           o   o    o    o     -   -
   Blocksize             R           o   -    o    o     -   -
   Cache-Control                r    -   -    o    -     -   -
   Connection                        o   o    o    o     o   o
   Content-Base          r           o   -    -    -     -   -
   Content-Base         4xx          o   o    o    o     o   o
   Content-Encoding      R      r    -   -    -    -     -   -
   Content-Encoding      r      r    o   -    -    -     -   -
   Content-Encoding     4xx     r    o   o    o    o     o   o
   Content-Language      R      r    -   -    -    -     -   -
   Content-Language      r      r    o   -    -    -     -   -
   Content-Language     4xx     r    o   o    o    o     o   o
   Content-Length        r      r    *   -    -    -     -   -
   Content-Length       4xx     r    *   *    *    *     *   *
   Content-Location      r           o   -    -    -     -   -
   Content-Location     4xx          o   o    o    o     o   o
   Content-Type          r           *   -    -    -     -   -
   Content-Type         4xx          *   *    *    *     *   *
   CSeq                 Rc           m   m    m    m     m   m
   Date                        am    o   o    o    o     o   o
   ETag                  r      r    o   -    o    -     -   -
   Expires               r      r    o   -    -    -     -   -
   From                  R      r    o   o    o    o     o   o
   Host                              -   -    -    -     -   -
   If-Match              R      r    -   -    o    -     -   -
   If-Modified-Since     R      r    o   -    o    -     -   -
   If-None-Match         R      r    o   -    -    -     -   -
   Last-Modified         r      r    o   -    -    -     -   -
   Location             3rr          o   o    o    o     o   o


   Table 9: Overview of RTSP header  fields  (A-L)  related  to  methods
   DESCRIBE, OPTIONS, SETUP, PLAY, PAUSE, and TEARDOWN.




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   Header              Where  Proxy DES OPT SETUP PLAY PAUSE TRD
   _____________________________________________________________
   Proxy-Authenticate   407    amr   m   m    m    m     m    m
   Proxy-Require         R     ar    o   o    o    o     o    o
   Public                r    admr   -  m*    -    -     -    -
   Public               501   admr  m*  m*   m*    m*   m*   m*
   Range                 R           -   -    -    o     o    -
   Range                 r           -   -    c    m*   m*    -
   Referer               R           o   o    o    o     o    o
   Require               R           o   o    o    o     o    o
   Retry-After        3rr,503        o   o    o    -     -    -
   RTP-Info              r           -   -    o    c     -    -
   Scale                             -   -    -    o     -    -
   Session               R           -   o    o    m     m    m
   Session               r           -   c    m    m     m    o
   Server                R           -   o    -    -     -    -
   Server                r           o   o    o    o     o    o
   Speed                             -   -    -    o     -    -
   Supported             R           o   o    o    o     o    o
   Supported             r           c   c    c    c     c    c
   Timestamp             R           o   o    o    o     o    o
   Timestamp             c           m   m    m    m     m    m
   Transport                         -   -    m    -     -    -
   Unsupported           r           c   c    c    c     c    c
   User-Agent            R          m*  m*   m*    m*   m*   m*
   Vary                  r           c   c    c    c     c    c
   Via                   R     amr   o   o    o    o     o    o
   Via                   c     dr    m   m    m    m     m    m
   WWW-Authenticate     401          m   m    m    m     m    m

   _____________________________________________________________
   Header              Where  Proxy DES OPT SETUP PLAY PAUSE TRD



   Table 10: Overview of RTSP header fields  (P-W)  related  to  methods
   DESCRIBE, OPTIONS, SETUP, PLAY, PAUSE, and TEARDOWN.


   any next hop that need to be approved by the requestor. See section    |
   17 for the usage of this header. It SHALL only be included in client   |
   to server requests, and server to client responses.                    |

   In a request the header SHALL contain the method (User, Proxy, or      |
   Any) for approving credentials selected by the requestor. The method   |
   SHALL NOT be changed by any proxy. If the method is "User" the header  |
   contains zero or more of credentials that the client accept. Each      |
   credential SHALL consist of one URI identifying the proxy or server,   |



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   Header              Where  Proxy GPR SPR RDR PNG
   __________________________________________________
   Accept-Credentials    R      r    o   o   o   o
   Accept-Credentials 470,407  ar    o   o   o   o
   Allow                405          m   m   m   m
   Authorization         R           o   o   o   o
   Bandwidth             R           -   o   -   -
   Blocksize             R           -   o   -   -
   Connection                        o   o   o   -
   Content-Base          R           o   o   -   -
   Content-Base          r           o   o   -   -
   Content-Base         4xx          o   o   o   -
   Content-Encoding      R      r    o   o   -   -
   Content-Encoding      r      r    o   o   -   -
   Content-Encoding     4xx     r    o   o   o   -
   Content-Language      R      r    o   o   -   -
   Content-Language      r      r    o   o   -   -
   Content-Language     4xx     r    o   o   o   -
   Content-Length        R      r    *   *   -   -
   Content-Length        r      r    *   *   -   -
   Content-Length       4xx     r    *   *   *   -
   Content-Location      R           o   o   -   -
   Content-Location      r           o   o   -   -
   Content-Location     4xx          o   o   o   -
   Content-Type          R           *   *   -   -
   Content-Type          r           *   *   -   -
   Content-Type         4xx          *   *   *   -
   CSeq                 Rc           m   m   m   m
   Date                        am    o   o   o   o
   From                  R      r    o   o   o   o
   Host                              -   -   -   -
   Last-Modified         R      r    -   -   -   -
   Last-Modified         r      r    o   -   -   -
   Location             3rr          o   o   o   o
   Location              R           -   -   m   -
   Proxy-Authenticate   407    amr   m   m   m   m
   Proxy-Require         R     ar    o   o   o   o
   Public               501   admr  m*  m*  m*  m*

   __________________________________________________
   Header              Where  Proxy GPR SPR RDR PNG


   Table 11: Overview of RTSP header fields  (A-P)  related  to  methods
   GET_PARAMETER, SET_PARAMETER, REDIRECT, and PING.






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             Header            Where  Proxy GPR SPR RDR PNG
             ________________________________________________
             Range               R           -   -   o   -
             Referer             R           o   o   o   -
             Require             R           o   o   o   o
             Retry-After      3rr,503        o   o   -   -
             Scale                           -   -   -   -
             Session             R           o   o   o   m
             Session             r           c   c   o   m
             Server              R           o   o   o   o
             Server              r           o   o   -   o
             Supported           R           o   o   o   o
             Supported           r           c   c   c   c
             Timestamp           R           o   o   o   o
             Timestamp           c           m   m   m   m
             Unsupported         r           c   c   c   c
             User-Agent          R          m*  m*   -  m*
             User-Agent          r           -   -  m*   -
             Vary                r           c   c   -   -
             Via                 R     amr   o   o   o   o
             Via                 c     dr    m   m   m   m
             WWW-Authenticate   401          m   m   m   m

             ________________________________________________
             Header            Where  Proxy GPR SPR RDR PNG


   Table 12: Overview of RTSP header fields  (R-W)  related  to  methods
   GET_PARAMETER, SET_PARAMETER, REDIRECT, and PING.


   and the SHA-1  [16] hash computed over that entity's DER encoded       |
   certificate [17] in Base64 [38].                                       |

   The Accept-Credentials header in a RTSP response SHALL, if included,   |
   contain the credentials information of the next hop that an            |
   intermediary needs to securely connect to. The credential MUST         |
   include the URI of the next proxy or server and the DER encoded        |
   X.509v3  [17] certificate in base64 [38].                              |


   Request Example:                                                       |
     Accept-Credentials:User,                                             |
          "rtsps://proxy2.example.com/";exaIl9VMbQMOFGClx5rXnPJKVNI=,     |
          "rtsps://server.example.com/";lurbjj5khhB0NhIuOXtt4bBRH1M=      |

   Response Example:                                                      |
     Accept-Credentials:"rtsps://proxy2.example.com/";XmW+39x4XdTLp...    |




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14.3 Accept-Encoding

   See [H14.3]

14.4 Accept-Language

   See [H14.4]. Note that the language specified applies to the
   presentation description and any reason phrases, not the media
   content.

14.5 Accept-Ranges

   The Accept-Ranges response-header field allows the server to indicate
   its acceptance of range requests and possible formats for a resource:


   Accept-Ranges: NPT, SMPTE



   This header has the same syntax as [H14.5] and the syntax is defined
   in 18.2.3. However new range-units are defined. Inclusion of any of
   the time formats indicates acceptance by the server for PLAY and
   PAUSE requests with this format. The headers value is valid for the
   resource specified by the URL in the request, this response
   corresponds to. A server SHOULD use this header in SETUP responses to
   indicate to the client which range time formats the media supports.
   The header SHOULD also be included in "456" responses which is a
   result of use of unsupported range formats.

14.6 Allow

   The Allow entity-header field lists the methods supported by the
   resource identified by the request-URL. The purpose of this field is
   to strictly inform the recipient of valid methods associated with the
   resource. An Allow header field MUST be present in a 405 (Method Not
   Allowed) response. See [H14.7] for syntax definition.

   Example of use:

     Allow: SETUP, PLAY, SET_PARAMETER



14.7 Authorization

   See [H14.8]




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14.8 Bandwidth

   The Bandwidth request-header field describes the estimated bandwidth
   available to the client, expressed as a positive integer and measured
   in bits per second. The bandwidth available to the client may change
   during an RTSP session, e.g., due to modem retraining.

   Example:

     Bandwidth: 4000



14.9 Blocksize

   The Blocksize request-header field is sent from the client to the
   media server asking the server for a particular media packet size.
   This packet size does not include lower-layer headers such as IP,
   UDP, or RTP. The server is free to use a blocksize which is lower
   than the one requested. The server MAY truncate this packet size to
   the closest multiple of the minimum, media-specific block size, or
   override it with the media-specific size if necessary. The block size
   MUST be a positive decimal number, measured in octets. The server
   only returns an error (4xx) if the value is syntactically invalid.

14.10 Cache-Control

   The Cache-Control general-header field is used to specify directives
   that MUST be obeyed by all caching mechanisms along the
   request/response chain.

   Cache directives must be passed through by a proxy or gateway
   application, regardless of their significance to that application,
   since the directives may be applicable to all recipients along the
   request/response chain. It is not possible to specify a cache-
   directive for a specific cache.

   Cache-Control should only be specified in a SETUP request and its
   response. Note: Cache-Control does not govern the caching of
   responses as for HTTP, instead it applies to the media stream
   identified by the SETUP request.  The caching of RTSP requests are
   generally not cacheable, for further information see 15. Below is the
   description of the cache directives that can be included in the
   Cache-Control header.

        no-cache: Indicates that the media stream MUST NOT be cached
             anywhere. This allows an origin server to prevent caching
             even by caches that have been configured to return stale



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             responses to client requests.

        public: Indicates that the media stream is cacheable by any
             cache.

        private: Indicates that the media stream is intended for a
             single user and MUST NOT be cached by a shared cache. A
             private (non-shared) cache may cache the media stream.

        no-transform: An intermediate cache (proxy) may find it useful
             to convert the media type of a certain stream. A proxy
             might, for example, convert between video formats to save
             cache space or to reduce the amount of traffic on a slow
             link.  Serious operational problems may occur, however,
             when these transformations have been applied to streams
             intended for certain kinds of applications. For example,
             applications for medical imaging, scientific data analysis
             and those using end-to-end authentication all depend on
             receiving a stream that is bit-for-bit identical to the
             original media stream. Therefore, if a response includes
             the no-transform directive, an intermediate cache or proxy
             MUST NOT change the encoding of the stream.  Unlike HTTP,
             RTSP does not provide for partial transformation at this
             point, e.g., allowing translation into a different
             language.

        only-if-cached: In some cases, such as times of extremely poor
             network connectivity, a client may want a cache to return
             only those media streams that it currently has stored, and
             not to receive these from the origin server. To do this,
             the client may include the only-if-cached directive in a
             request. If it receives this directive, a cache SHOULD
             either respond using a cached media stream that is
             consistent with the other constraints of the request, or
             respond with a 504 (Gateway Timeout) status. However, if a
             group of caches is being operated as a unified system with
             good internal connectivity, such a request MAY be forwarded
             within that group of caches.

        max-stale: Indicates that the client is willing to accept a
             media stream that has exceeded its expiration time. If
             max-stale is assigned a value, then the client is willing
             to accept a response that has exceeded its expiration time
             by no more than the specified number of seconds. If no
             value is assigned to max-stale, then the client is willing
             to accept a stale response of any age.

        min-fresh: Indicates that the client is willing to accept a



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             media stream whose freshness lifetime is no less than its
             current age plus the specified time in seconds. That is,
             the client wants a response that will still be fresh for at
             least the specified number of seconds.

        must-revalidate: When the must-revalidate directive is present
             in a SETUP response received by a cache, that cache MUST
             NOT use the entry after it becomes stale to respond to a
             subsequent request without first revalidating it with the
             origin server.  That is, the cache must do an end-to-end
             revalidation every time, if, based solely on the origin
             server's Expires, the cached response is stale.)

        proxy-revalidate: The proxy-revalidate directive has the same
             meaning as the must-revalidate directive, except that it
             does not apply to non-shared user agent caches. It can be
             used on a response to an authenticated request to permit
             the user's cache to store and later return the response
             without needing to revalidate it (since it has already been
             authenticated once by that user), while still requiring
             proxies that service many users to revalidate each time (in
             order to make sure that each user has been authenticated).
             Note that such authenticated responses also need the public
             cache control directive in order to allow them to be cached
             at all.

        max-age: When an intermediate cache is forced, by means of a
             max-age=0 directive, to revalidate its own cache entry, and
             the client has supplied its own validator in the request,
             the supplied validator might differ from the validator
             currently stored with the cache entry. In this case, the
             cache MAY use either validator in making its own request
             without affecting semantic transparency.

             However, the choice of validator might affect performance.
             The best approach is for the intermediate cache to use its
             own validator when making its request. If the server
             replies with 304 (Not Modified), then the cache can return
             its now validated copy to the client with a 200 (OK)
             response. If the server replies with a new entity and cache
             validator, however, the intermediate cache can compare the
             returned validator with the one provided in the client's
             request, using the strong comparison function. If the
             client's validator is equal to the origin server's, then
             the intermediate cache simply returns 304 (Not Modified).
             Otherwise, it returns the new entity with a 200 (OK)
             response.




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14.11 Connection

   See [H14.10]. The use of the connection option "close" in RTSP
   messages SHOULD be limited to error messages when the server is
   unable to recover and therefore see it necessary to close the
   connection. The reason is that the client shall have the choice of
   continue using a connection indefinitely as long as it sends valid
   messages.

14.12 Content-Base

   The Content-Base entity-header field may be used to specify the base
   URL for resolving relative URLs within the entity.

   Content-Base: rtsp://media.example.com/movie/twister


   If no Content-Base field is present, the base URL of an entity is
   defined either by its Content-Location (if that Content-Location URL
   is an absolute URL) or the URL used to initiate the request, in that
   order of precedence. Note, however, that the base URL of the contents
   within the entity-body may be redefined within that entity-body.

14.13 Content-Encoding

   See [H14.11]

14.14 Content-Language

   See [H14.12]

14.15 Content-Length

   The Content-Length general-header field contains the length of the
   content of the method (i.e. after the double CRLF following the last
   header). Unlike HTTP, it MUST be included in all messages that carry
   content beyond the header portion of the message. If it is missing, a
   default value of zero is assumed. It is interpreted according to
   [H14.13].

14.16 Content-Location

   See [H14.14]

14.17 Content-Type

   See [H14.17]. Note that the content types suitable for RTSP are
   likely to be restricted in practice to presentation descriptions and



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   parameter-value types.

14.18 CSeq

   The CSeq general-header field specifies the sequence number for an
   RTSP request-response pair. This field MUST be present in all
   requests and responses. For every RTSP request containing the given
   sequence number, the corresponding response will have the same
   number. Any retransmitted request must contain the same sequence
   number as the original (i.e. the sequence number is not incremented
   for retransmissions of the same request). For each new RTSP request
   the CSeq value SHALL be incremented by one. The initial sequence
   number MAY be any number, however it is RECOMMENDED to start at 1.
   Each sequence number series is unique between each requester and
   responder, i.e. the client has one series for its request to a server
   and the server has another when sending request to the client. Each
   requester and responder is identified with its network address.

   Example:

   CSeq: 239



14.19 Date

   See [H14.18]. An RTSP message containing a body MUST include a Date
   header if the sending host has a clock. Servers SHOULD include a Date
   header in all other RTSP messages.


14.20 ETag                                                                |

   The ETag response header MAY be included in DESCRIBE or SETUP          |
   responses. The entity tag returned in a DESCRIBE response is for the   |
   included entity, while for SETUP it refers to the media resource just  |
   set up. This differentiation allows for cache validation of both       |
   session description and the media resource associated with the         |
   description. If the ETag is provided both inside the entity, e.g.      |
   within the "a=etag" attribute in SDP, and in the response message,     |
   then both tags SHALL be identical. It is RECOMMENDED that the ETag is  |
   primarily given in the RTSP response message, to ensure that caches    |
   can use the ETag without requiring content inspection.                 |

   SETUP and DESCRIBE requests can be made conditional upon the ETag      |
   using the headers If-Match (Section 14.24) and If-None-Match (Section  |
   14.26).                                                                |




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14.21 Expires

   The Expires entity-header field gives a date and time after which the
   description or media-stream should be considered stale. The
   interpretation depends on the method:

        DESCRIBE response: The Expires header indicates a date and time
             after which the description SHOULD be considered stale.

        SETUP response: The Expires header indicate a date and time
             after which the media stream SHOULD be considered stale.

   A stale cache entry may not normally be returned by a cache (either a
   proxy cache or an user agent cache) unless it is first validated with
   the origin server (or with an intermediate cache that has a fresh
   copy of the entity). See section 15 for further discussion of the
   expiration model.

   The presence of an Expires field does not imply that the original
   resource will change or cease to exist at, before, or after that
   time.

   The format is an absolute date and time as defined by HTTP-date in
   [H3.3]; it MUST be in RFC1123-date format:

   An example of its use is

     Expires: Thu, 01 Dec 1994 16:00:00 GMT



   RTSP/1.0 clients and caches MUST treat other invalid date formats,
   especially including the value "0", as having occurred in the past
   (i.e., already expired).

   To mark a response as "already expired," an origin server should use
   an Expires date that is equal to the Date header value. To mark a
   response as "never expires," an origin server SHOULD use an Expires
   date approximately one year from the time the response is sent.
   RTSP/1.0 servers SHOULD NOT send Expires dates more than one year in
   the future.

   The presence of an Expires header field with a date value of some
   time in the future on a media stream that otherwise would by default
   be non-cacheable indicates that the media stream is cacheable, unless
   indicated otherwise by a Cache-Control header field (Section 14.10).

14.22 From



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   See [H14.22].

14.23 Host

   The Host HTTP request header field [H14.23] is not needed for RTSP,
   and SHALL NOT be sent. It SHALL be silently ignored if received.

14.24 If-Match

   See [H14.24].  The If-Match request-header field is especially useful  |
   for ensuring the integrity of the presentation description, in both    |
   the case where it is fetched via means external to RTSP (such as       |
   HTTP), or in the case where the server implementation is guaranteeing  |
   the integrity of the description between the time of the DESCRIBE      |
   message and the SETUP message. By including the ETag given in or with  |
   the session description in a SETUP request, the client ensures that    |
   resources set up are matching the description. A SETUP request for     |
   which the ETag validation check fails, SHALL responde using 412        |
   (Precondition Failed).                                                 |

   This validation check is also very useful if a session has been        |
   redirected from one server to another.                                 |


14.25 If-Modified-Since

   The If-Modified-Since request-header field is used with the DESCRIBE
   and SETUP methods to make them conditional. If the requested variant
   has not been modified since the time specified in this field, a
   description will not be returned from the server (DESCRIBE) or a
   stream will not be set up (SETUP). Instead, a 304 (Not Modified)
   response SHALL be returned without any message-body.

   An example of the field is:

     If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT



14.26 If-None-Match

   See [H14.26].

   This request header can be used with entity tags to make DESCRIBE
   requests conditional. A new session description is retrieved only if
   another entity than the already available would be included. If the
   entity available for delivery is matching the one the client already
   has, then a 304 (Not Modified) response is given.



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14.27 Last-Modified

   The Last-Modified entity-header field indicates the date and time at
   which the origin server believes the presentation description or
   media stream was last modified. See [H14.29]. For the methods
   DESCRIBE, the header field indicates the last modification date and
   time of the description, for SETUP that of the media stream.

14.28 Location

   See [H14.30].

14.29 Proxy-Authenticate

   See [H14.33].

14.30 Proxy-Require

   The Proxy-Require request-header field is used to indicate proxy-
   sensitive features that MUST be supported by the proxy. Any Proxy-
   Require header features that are not supported by the proxy MUST be
   negatively acknowledged by the proxy to the client using the
   Unsupported header. The proxy SHALL use the 551 (Option Not
   Supported) status code in the response. Any feature tag included in
   the Proxy-Require does not apply to the server. To ensure that a
   feature is supported by both proxies and servers the tag must be
   included in also a Require header.

   See Section 14.35 for more details on the mechanics of this message
   and a usage example.

   Example of use:

      Proxy-Require: play.basic



14.31 Public

   The Public response header field lists the set of methods supported    |
   by the response sender. This header applies to the general             |
   capabilities of the sender and its only purpose is to indicate the     |
   sender's capabilities to the recipient. The methods listed may or may
   not be applicable to the Request-URL; the Allow header field (section
   14.7) MAY be used to indicate methods allowed for a particular URL.

   Example of use:




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      Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN



   In the event that there are proxies between the sender and the         |
   recipient of a response, each intervening proxy MUST modify the        |
   Public header field to remove any methods that are not supported via   |
   that proxy. The resulting Public header field will contain an          |
   intersection of the sender's methods and the methods allowed through   |
   by the intervening proxies.                                            |

        In general proxies should allow all methods to               |
        transparently pass through from the sending RTSP agent to    |
        the receiving RTSP agent, but there may be cases where this  |
        is not desirable for a given proxy. Modification of the      |
        Public response header field by the intervening proxies      |
        ensures that the request sender gets an accurate response    |
        indicating the methods that can be used on the target agent  |
        via the proxy chain.

14.32 Range

   The Range request and response header field specifies a range of
   time. The header is used in PLAY request to indicate the range of
   time the client desires the server to play back. The Range header in
   a PLAY indicates what range of time that is actually being played. In
   a SETUP response the header MAY be used, to ensure correct
   synchronization information after changes of transport parameters.

   The range can be specified in a number of units. This specification
   defines the smpte (Section 3.4), npt (Section 3.5), and clock
   (Section 3.6) range units. Within RTSP, byte ranges [H14.35.1] are
   normally not meaningful, and the behavior is unspecified, but they
   and other extended units MAY be used. Servers supporting the Range
   header MUST understand the NPT range format and SHOULD understand the
   SMPTE range format. If the Range header is given in a time format
   that is not understood, the recipient should return 456 (Header Field
   Not Valid for Resource) and include a Accept-Ranges header indicating
   which time format that is supported for this resource.

   The header MAY contain a time parameter in UTC, specifying the time
   at which the operation is to be made effective. This functionality
   SHALL only be used with the REDIRECT method.

   Ranges are half-open intervals, including the first point, but
   excluding the second point. In other words, a range of A-B starts
   exactly at time A, but stops just before B. Only the start time of a
   media unit such as a video or audio frame is relevant. As an example,



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   assume that video frames are generated every 40 ms. A range of
   10.0-10.1 would include a video frame starting at 10.0 or later time
   and would include a video frame starting at 10.08, even though it
   lasted beyond the interval. A range of 10.0-10.08, on the other hand,
   would exclude the frame at 10.08.

   Example:

     Range: clock=19960213T143205Z-;time=19970123T143720Z



        The notation is similar to that used for the HTTP/1.1 [4]
        byte-range header. It allows clients to select an excerpt
        from the media object, and to play from a given point to
        the end as well as from the current location to a given
        point. The start of playback can be scheduled for any time
        in the future, although a server may refuse to keep server
        resources for extended idle periods.

   By default, range intervals increase, where the second point is
   larger than the first point.

   Example:

       Range: npt=10-15



   However, range intervals can also decrease if the Scale header (see
   section  14.37) indicates a negative scale value. For example, this
   would be the case when a playback in reverse is desired.

   Example:

       Scale: -1
       Range: npt=15-10



   Decreasing ranges are still half open intervals as described above.
   Thus, For range A-B, A is closed and B is open. In the above example,
   15 is closed and 10 is open. An exception to this rule is the case
   when B=0 in a decreasing range. In this case, the range is closed on
   both ends, as otherwise there would be no way to reach 0 on a reverse
   playback.

   Example:



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       Scale: -1
       Range: npt=15-0



   In this range both 15 and 0 are closed.

   A decreasing range interval without a corresponding negative Scale
   header is not valid.

14.33 Referer

   See [H14.36]. The URL refers to that of the presentation description,
   typically retrieved via HTTP.

14.34 Retry-After

   See [H14.37].

14.35 Require

   The Require request-header field is used by clients or servers to
   ensure that the other end-point supports features that are required
   in respect to this request. It can also be used to query if the other
   end-point supports certain features, however the use of the Supported
   (Section  14.41) is much more effective in this purpose. The server
   MUST respond to this header by using the Unsupported header to
   negatively acknowledge those feature-tags which are NOT supported.
   The response SHALL use the error code 551 (Option Not Supported).
   This header does not apply to proxies, for the same functionality in
   respect to proxies see, header Proxy-Require (Section 14.30).


        This is to make sure that the client-server interaction
        will proceed without delay when all features are understood
        by both sides, and only slow down if features are not
        understood (as in the example below). For a well-matched
        client-server pair, the interaction proceeds quickly,
        saving a round-trip often required by negotiation
        mechanisms. In addition, it also removes state ambiguity
        when the client requires features that the server does not
        understand.

   Example:

   C->S:   SETUP rtsp://server.com/foo/bar/baz.rm RTSP/1.0
           CSeq: 302
           Require: funky-feature



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           Funky-Parameter: funkystuff

   S->C:   RTSP/1.0 551 Option not supported
           CSeq: 302
           Unsupported: funky-feature




   In this example, "funky-feature" is the feature-tag which indicates
   to the client that the fictional Funky-Parameter field is required.
   The relationship between "funky-feature" and Funky-Parameter is not
   communicated via the RTSP exchange, since that relationship is an
   immutable property of "funky-feature" and thus should not be
   transmitted with every exchange.

   Proxies and other intermediary devices SHOULD ignore features that
   are not understood in this field. If a particular extension requires
   that intermediate devices support it, the extension should be tagged
   in the Proxy-Require field instead (see Section 14.30).

14.36 RTP-Info

   The RTP-Info response-header field is used to set RTP-specific         |
   parameters in the PLAY response. These parameters correspond to the    |
   synchronization source identified by the first value of the ssrc       |
   parameter of the Transport response header in the SETUP response. For  |
   streams using RTP as transport protocol the RTP-Info header SHOULD be  |
   part of a 200 response to PLAY.


        The exclusion of the RTP-Info in a PLAY response for RTP
        transported media will result in that a client needs to
        synchronize the media streams using RTCP. This may have
        negative impact as the RTCP can be lost, and does not need
        to be particulary timely in their arrival. Also
        functionality as informing the client from which packet a
        seek has occurred is affected.

   The RTP-Info MAY also be included in SETUP responses to provide
   synchronization information when changing transport parameters, see
   11.3.

   The header can carry the following parameters:

        url: Indicates the stream URL which for which the following RTP
             parameters correspond, this URL MUST be the same used in
             the SETUP request for this media stream. Any relative URL



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             SHALL use the request URL as base URL.


        seq: Indicates the sequence number of the first packet of the     |
             stream that is direct result of the request.  This allows    |
             clients to gracefully deal with packets when seeking.  The   |
             client uses this value to differentiate packets that         |
             originated before the seek from packets that originated      |
             after the seek. Note that a client may not receive the       |
             packet with the expressed sequence number, and instead       |
             packets with a higher sequence number, due to packet loss    |
             or reordering.

        rtptime: Indicates the RTP timestamp corresponding to the time
             value in the Range response header. (Note: For aggregate
             control, a particular stream may not actually generate a
             packet for the Range time value returned or implied. Thus,
             there is no guarantee that the packet with the sequence
             number indicated by seq actually has the timestamp
             indicated by rtptime.) The client uses this value to
             calculate the mapping of RTP time to NPT.


             A mapping from RTP timestamps to NTP timestamps (wall
             clock) is available via RTCP. However, this
             information is not sufficient to generate a mapping
             from RTP timestamps to NPT. Furthermore, in order to
             ensure that this information is available at the
             necessary time (immediately at startup or after a
             seek), and that it is delivered reliably, this mapping
             is placed in the RTSP control channel.

             In order to compensate for drift for long, uninterrupted
             presentations, RTSP clients should additionally map NPT to
             NTP, using initial RTCP sender reports to do the mapping,
             and later reports to check drift against the mapping.

   Additionally, the RTP-Info header parameter fields only apply to a     |
   single SSRC within a stream (the first SSRC reported in the transport  |
   response header; see section  14.43). If there are multiple            |
   synchronization sources (SSRCs) present within a RTP session           |
   transmitting media, RTCP must be used to map RTP and NTP timestamps    |
   for those sources, for both synchronization and drift-checking. Due    |
   to backwards compatibility reasons these shortcomings can't be fixed   |
   without defining a new header, which is for future work if needed.

   Additional constraint: The syntax element "safe-url" (see section
   18.2.3) MUST NOT contain the semicolon (";") or comma (",")



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   characters. The quoted-url form SHOULD only be used when a URL does
   not meet the safe-url constraint, in order to ensure compatibility
   with implementations conformant to RFC 2326 [1].

   Example:

   RTP-Info: url=rtsp://example.com/bar.avi/streamid=0;seq=45102,
             url=rtsp://example.com/bar.avi/streamid=1;seq=30211



14.37 Scale

   A scale value of 1 indicates normal play at the normal forward
   viewing rate. If not 1, the value corresponds to the rate with
   respect to normal viewing rate. For example, a ratio of 2 indicates
   twice the normal viewing rate ("fast forward") and a ratio of 0.5
   indicates half the normal viewing rate. In other words, a ratio of 2
   has normal play time increase at twice the wallclock rate. For every
   second of elapsed (wallclock) time, 2 seconds of content will be
   delivered. A negative value indicates reverse direction. For certain
   media transports this may require certain considerations to work
   consistent, see section B.1 for description on how RTP handles this.

   Unless requested otherwise by the Speed parameter, the data rate
   SHOULD not be changed. Implementation of scale changes depends on the
   server and media type. For video, a server may, for example, deliver
   only key frames or selected key frames. For audio, it may time-scale
   the audio while preserving pitch or, less desirably, deliver
   fragments of audio.

   The server should try to approximate the viewing rate, but may
   restrict the range of scale values that it supports. The response
   MUST contain the actual scale value chosen by the server.

   If the server does not implement the possibility to scale, it will
   not return a Scale header. A server supporting Scale operations for
   PLAY SHALL indicate this with the use of the "play.scale" feature-
   tags.

   When indicating a negative scale for a reverse playback, the Range
   header must indicate a decreasing range as described in section
   14.32.

   Example of playing in reverse at 3.5 times normal rate:

     Scale: -3.5
     Range: npt=15-10



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14.38 Speed

   The Speed request-header field requests the server to deliver data to
   the client at a particular speed, contingent on the server's ability
   and desire to serve the media stream at the given speed.
   Implementation by the server is OPTIONAL. The default is the bit rate
   of the stream.

   The parameter value is expressed as a decimal ratio, e.g., a value of
   2.0 indicates that data is to be delivered twice as fast as normal. A
   speed of zero is invalid. All speeds may not be possible to support.
   Therefore the actual used speed MUST be included in the response. The
   lack of a response header is indication of lack of support from the
   server of this functionality. Support of the speed functionality are
   indicated by the "play.speed" featuretag.

   Example:

     Speed: 2.5



   Use of this field changes the bandwidth used for data delivery. It is
   meant for use in specific circumstances where preview of the
   presentation at a higher or lower rate is necessary. Implementors
   should keep in mind that bandwidth for the session may be negotiated
   beforehand (by means other than RTSP), and therefore re-negotiation
   may be necessary. When data is delivered over UDP, it is highly
   recommended that means such as RTCP be used to track packet loss
   rates. If the data transport is performed over public best-effort
   networks the sender SHOULD perform congestion control of the
   stream(s). This can result in that the communicated speed is
   impossible to maintain.

14.39 Server

   See [H14.38], however the header syntax is corrected in section
   18.2.3.

14.40 Session

   The Session request-header and response-header field identifies an
   RTSP session. An RTSP session is created by the server as a result of
   a successful SETUP request and in the response the session identifier
   is given to the client. The RTSP session exist until destroyed by a
   TEARDOWN or timed out by the server.

   The session identifier is chosen by the server (see Section 3.3) and



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   MUST be returned in the SETUP response. Once a client receives a
   session identifier, it SHALL be included in any request related to
   that session.  This means that the Session header MUST be included in
   a request using the following methods: PLAY, PAUSE, PING, and
   TEARDOWN, and MAY be included in SETUP, OPTIONS, SET_PARAMETER,
   GET_PARAMETER, and REDIRECT, and SHALL NOT be included in DESCRIBE.
   In a RTSP response the session header SHALL be included in methods,
   SETUP, PING, PLAY, and PAUSE, and MAY be included in methods,
   TEARDOWN, and REDIRECT, and if included in the request of the
   following methods it SHALL also be included in the response, OPTIONS,
   GET_PARAMETER, and SET_PARAMETER, and SHALL NOT be included in
   DESCRIBE.

   Note that RFC 2326 servers and client may in some cases not include
   or return a Session header when expected according to the above text.
   Any client or server is RECOMMENDED to be forgiving of this error if
   possible (which it is in many cases).

   The timeout parameter MAY be included in a SETUP response, and SHALL
   NOT be included in requests. The server uses it to indicate to the
   client how long the server is prepared to wait between RTSP commands
   or other signs of life before closing the session due to lack of
   activity (see below and Section A). The timeout is measured in
   seconds, with a default of 60 seconds (1 minute). The length of the
   session timeout SHALL NOT be changed in a established session.

   The mechanisms for showing liveness of the client is, any RTSP
   request with a Session header, if RTP & RTCP is used an RTCP message,
   or through any other used media protocol capable of indicating
   liveness of the RTSP client. It is RECOMMENDED that a client does not
   wait to the last second of the timeout before trying to send a
   liveness message. The RTSP message may be lost or when using reliable
   protocols, such as TCP, the message may take some time to arrive
   safely at the receiver. To show liveness between RTSP request issued
   to accomplish other things, the following mechanisms can be used, in
   descending order of preference:

        RTCP: If RTP is used for media transport RTCP SHOULD be used. If
             RTCP is used to report transport statistics, it SHALL also
             work as keep alive. The server can determine the client by
             used network address and port together with the fact that
             the client is reporting on the servers SSRC(s). A downside
             of using RTCP is that it only gives statistical guarantees
             to reach the server. However that probability is so low
             that it can be ignored in most cases. For example, a
             session with 60 seconds timeout and enough bitrate assigned
             to RTCP messages to send a message from client to server on
             average every 5 seconds. That client have for a network



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             with 5 % packet loss, the probability to fail showing
             liveness sign in that session within the timeout interval
             of 2.4*E-16. In sessions with shorter timeout times, or
             much higher packet loss, or small RTCP bandwidths SHOULD
             also use any of the mechanisms below.

        PING: The use of the PING method is the best of the RTSP based
             methods. It has no other effects than updating the timeout
             timer. In that way it will be a minimal message, that also
             does not cause any extra processing for the server. The
             downside is that it may not be implemented. A client SHOULD
             use a OPTIONS request to verify support of the PING at the
             server. It is also possible to detect support by sending a
             PING to the server. If a 200 (OK) message is received the
             server supports it. In case a 501 (Not Implemented) is
             received it does not support PING and there is no meaning
             in continue trying.  Also the reception of a error message
             will also mean that the liveness timer has not been
             updated.

        SET_PARAMETER: When using SET_PARAMETER for keep alive, no body
             SHOULD be included. This method is basically as good as
             PING, however the implementation support of the method is
             today limited. The same considerations as for PING apply
             regarding checking of support in server and proxies.

        OPTIONS: This method does also work. However it causes the
             server to perform unnecessary processing and result in
             bigger responses than necessary for the task. The reason
             for this is that the Public is always included creating
             overhead.

   Note that a session identifier identifies an RTSP session across
   transport sessions or connections. RTSP requests for a given session
   can use different URLs (Presentation and media URLs).  Note, that
   there are restrictions depending on the session which URLs that are
   acceptable for a given method. However, multiple "user" sessions for
   the same URL from the same client will require use of different
   session identifiers.

        The session identifier is needed to distinguish several
        delivery requests for the same URL coming from the same
        client.

   The response 454 (Session Not Found) SHALL be returned if the session
   identifier is invalid.

14.41 Supported



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   The Supported header field enumerates all the extensions supported by  |
   the client or server using feature tags. The header carries the        |
   extensions supported by the message sending entity.  The Supported     |
   header MAY be included in any request.  When present in a request,     |
   the receiver MUST respond with its corresponding Supported header.     |
   Note, also in 4xx and 5xx responses shall the supported header be      |
   included.

   The Supported header field contains a list of feature-tags, described
   in Section 3.7, that are understood by the client or server.

   Example:

     C->S:  OPTIONS rtsp://example.com/ RTSP/1.0
            Supported: foo, bar, blech

     S->C:  RTSP/1.0 200 OK
            Supported: bar, blech, baz



14.42 Timestamp

   The Timestamp general-header field describes when the client sent the
   request to the server. The value of the timestamp is of significance
   only to the client and may use any timescale. The server MUST echo
   the exact same value and MAY, if it has accurate information about
   this, add a floating point number indicating the number of seconds
   that has elapsed since it has received the request. The timestamp is
   used by the client to compute the round-trip time to the server so
   that it can adjust the timeout value for retransmissions. It also
   resolves retransmission ambiguities for unreliable transport of RTSP.

14.43 Transport

   The Transport request and response header field indicates which
   transport protocol is to be used and configures its parameters such
   as destination address, compression, multicast time-to-live and
   destination port for a single stream. It sets those values not
   already determined by a presentation description.

   Transports are comma separated, listed in order of preference.
   Parameters may be added to each transport, separated by a semicolon.
   The server SHOULD return a Transport response-header field in the
   response to indicate the values actually chosen. The Transport header
   field MAY also be used to change certain transport parameters. A
   server MAY refuse to change parameters of an existing stream.




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   A Transport request header field MAY contain a list of transport
   options acceptable to the client, in the form of multiple
   transportspec entries. In that case, the server MUST return the
   single option (transport-spec) which was actually chosen. The number
   of transportspec entries is expected to be limited as the client will
   get guidance on what configurations that are possible from the
   presentation description.

   A transport-spec transport option may only contain one of any given
   parameter within it. Parameters may be given in any order.
   Additionally, it may only contain the unicast or multicast transport
   parameter. Unknown transport parameters SHALL be ignored. The
   requester need to ensure that the responder understands the
   parameters through the use of feature tags and the Require header.

   The usage of any parameter that was not defined in RFC 2326 or in an
   extended way requires that request or response contains a Require
   header with the "play.basic" feature tag.


        The Transport header field is restricted to describing a
        single media stream. (RTSP can also control multiple
        streams as a single entity.) Making it part of RTSP rather
        than relying on a multitude of session description formats
        greatly simplifies designs of firewalls.

   The syntax for the transport specifier is

   transport/profile/lower-transport.


   The default value for the "lower-transport" parameters is specific to
   the profile. For RTP/AVP, the default is UDP.

   There is three different methods for how to specify where the media
   should be delivered:

        o The presence of this parameter and its values indicates
          address and port pairs for one or more IP flow necessary for
          the media transport. This is an improved version of the
          Destination parameter.

        o The presence of this parameter and its value indicates what IP
          address the media shall be delivered to. This method is kept
          for backwards compatibility reasons, dest_addr is a better
          choice.

        o The lack of of both of the above parameters indicates that the



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          server SHALL send media to same address for which the RTSP
          messages originates.

   The choice of method for indicating where the media shall be
   delivered depends on the use case. In many case the only allowed
   method will be to use no explicit indication and have the server
   deliver media to the source of the RTSP messages.

   An RTSP proxy will also need to take care. If the media is not
   desired to be routed through the proxy, the proxy will need to
   introduce the destination indication.

   Below are the configuration parameters associated with transport:

   General parameters:

        unicast / multicast: This parameter is a mutually exclusive
             indication of whether unicast or multicast delivery will be
             attempted. One of the two values MUST be specified. Clients
             that are capable of handling both unicast and multicast
             transmission MUST indicate such capability by including two
             full transport-specs with separate parameters for each.

        destination: The address of the stream recipient to which a
             stream will be sent. The client originating the RTSP
             request may specify the destination address of the stream
             recipient with the destination parameter. When the
             destination field is specified, the recipient may be a
             different party than the originator of the request. To
             avoid becoming the unwitting perpetrator of a remote-
             controlled denial-of-service attack, a server SHOULD
             authenticate the client originating the request and SHOULD
             log such attempts before allowing the client to direct a
             media stream to a recipient address not chosen by the
             server. While, this is particularly important if RTSP
             commands are issued via UDP, implementations cannot rely on
             TCP as reliable means of client identification by itself
             either.

             The server SHOULD NOT allow the destination field to be set
             unless a mechanism exists in the system to authorize the
             request originator to direct streams to the recipient. It
             is preferred that this authorization be performed by the
             recipient itself and the credentials passed along to the
             server. However, in certain cases, such as when recipient
             address is a multicast group, or when the recipient is
             unable to communicate with the server in an out-of-band
             manner, this may not be possible. In these cases server may



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             chose another method such as a server-resident
             authorization list to ensure that the request originator
             has the proper credentials to request stream delivery to
             the recipient.

             This parameter SHALL NOT be used when src_addr and
             dest_addr is used in a transport declaration. For IPv6
             addresses it is RECOMMENDED that they be given as fully
             qualified domain to make it backwards compatible with RFC
             2326 implementations.

        source: If the source address for the stream is different than
             can be derived from the RTSP endpoint address (the server
             in playback), the source address SHOULD be specified. To
             maintain backwards compatibility with RFC 2326, any IPv6
             host's address must be given as a fully qualified domain
             name. This parameter SHALL NOT be used when src_addr and
             dest_addr is used in a transport declaration.


             This information may also be available through SDP.
             However, since this is more a feature of transport
             than media initialization, the authoritative source
             for this information should be in the SETUP response.

        layers: The number of multicast layers to be used for this media
             stream. The layers are sent to consecutive addresses
             starting at the destination address.

        dest_addr: A general destination address parameter that can
             contain one or more address and port pair. For each
             combination of Protocol/Profile/Lower Transport the
             interpretation of the address or addresses needs to be
             defined.  The host address part of the tuple MAY be empty,
             for example ":8000", in cases when only destination port is
             desired to be specified.

             The client or server SHALL NOT use this parameter unless
             both client and server has shown support. This parameter
             MUST be supported by client and servers that implements
             this specification. Support is indicated by the use of the
             feature-tag "play.basic". This parameter SHALL NOT be used
             in the same transport specification as any of the
             parameters "destination", "source", "port", "client_port",
             and "server_port".

             The same security consideration that are given for the
             "Destination" parameter does also applies to this



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             parameter. This parameter can be used for redirecting
             traffic to recipient not desiring the media traffic.

        src_addr: A General source address parameter that can contain
             one or more address and port pair. For each combination of
             Protocol/Profile/Lower Transport the interpretation of the
             address or addresses needs to be defined. The client or
             server SHALL NOT use this parameter unless both client and
             server has shown support. This parameter MUST be supported
             by client and servers that implements this specification.
             Support is indicated by the use the feature-tag
             "play.basic". This parameter SHALL NOT be used in the same
             transport specification as any of the parameters
             "destination", "source", "port", "client_port", and
             "server_port".

             This parameter MUST be specified by the server if it
             transmits media packets from another address than the one
             RTSP messages are sent to. This will allow the client to
             verify source address and give it a destination address for
             its RTCP feedback packets if RTP is used. The address or
             addresses indicated in the src_addr parameter SHOULD be
             used both for sending and receiving of the media streams
             data packets. The main reasons are three: First by sending
             from the indicated ports the source address will be known
             by the receiver of the packet. Secondly, in the presence of
             NATs some traversal mechanism requires either knowledge
             from which address and port a packet flow is coming, or
             having the possibility to send data to the sender port.

        mode: The mode parameter indicates the methods to be supported
             for this session. Valid values are PLAY and RECORD. If not
             provided, the default is PLAY.  The RECORD value was
             defined in RFC 2326 and is deprecated in this
             specification.

        append: The append parameter was used together with RECORD and
             is now deprecated.

        interleaved: The interleaved parameter implies mixing the media
             stream with the control stream in whatever protocol is
             being used by the control stream, using the mechanism
             defined in Section 12. The argument provides the channel
             number to be used in the $ statement and MUST be present.
             This parameter MAY be specified as a range, e.g.,
             interleaved=4-5 in cases where the transport choice for the
             media stream requires it, e.g. for RTP with RTCP.  The
             channel number given in the request are only a guidance



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             from the client to the server on what channel number(s) to
             use. The server MAY set any valid channel number in the
             response. The declared channel(s) are bi-directional, so
             both end-parties MAY send data on the given channel. One
             example of such usage is the second channel used for RTCP,
             where both server and client sends RTCP packets on the same
             channel.


             This allows RTP/RTCP to be handled similarly to the
             way that it is done with UDP, i.e., one channel for
             RTP and the other for RTCP.

   Multicast-specific:

        ttl: multicast time-to-live.

   RTP-specific:

   These parameters are MAY only be used if the media transport protocol
   is RTP.

        port: This parameter provides the RTP/RTCP port pair for a
             multicast session. It is should be specified as a range,
             e.g., port=3456-3457

        client_port: This parameter provides the unicast RTP/RTCP port
             pair on the client where media data and control information
             is to be sent. It is specified as a range, e.g.,
             port=3456-3457. This parameter SHALL NOT be used when
             src_addr and dest_addr is used in a transport declaration.

        server_port: This parameter provides the unicast RTP/RTCP port
             pair on the server where media data and control information
             is to be sent. It is specified as a range, e.g.,
             port=3456-3457. This parameter SHALL NOT be used when
             src_addr and dest_addr is used in a transport declaration.


        ssrc: The ssrc parameter, if included in a SETUP response,        |
             indicates the RTP SSRC [18] value(s) that will be used by    |
             the media server for RTP packets within the stream. It is    |
             expressed as an eight digit hexadecimal value. Multiple      |
             values MAY only be specified if the cl¬ćent has indicated     |
             support for this specification, i.e.  if including multiple  |
             SSRC values, the request must have include "Require:         |
             play.basic" or "Supported: play.basic". If no such support   |
             is present only a single value SHALL be included.            |



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             If the server does not act as a synchronization source for   |
             stream data (for instance, server is a translator,           |
             reflector, etc.), and only a single value can be specified,  |
             the value will be the "packet sender's SSRC" that would      |
             have been used in the RTCP Receiver Reports generated by     |
             the server, regardless of whether the server actually        |
             generates RTCP RRs.                                          |

             The first SSRC value is the one that RTP-Info                |
             synchronization information relates to, see section  14.36.

             The functionality of specifying the ssrc parameter in a
             SETUP request is deprecated as it is incompatible with the
             specification of RTP in RFC 3550  [18]. If the parameter is
             included in the transport header of a SETUP request, the
             server MAY ignore it, and choose an appropriate SSRC for
             the stream. The server MAY set the ssrc parameter in the
             transport header of the response.

   The combination of transport protocol, profile and lower transport
   needs to be defined. A number of combinations are defined in the
   appendix B.

   Below is a usage example, showing a client advertising the capability
   to handle multicast or unicast, preferring multicast.  Since this is
   a unicast-only stream, the server responds with the proper transport
   parameters for unicast.


     C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.0
           CSeq: 302
           Transport: RTP/AVP;multicast;mode="PLAY",
               RTP/AVP;unicast;client_port=3456-3457;mode="PLAY"

     S->C: RTSP/1.0 200 OK
           CSeq: 302
           Date: 23 Jan 1997 15:35:06 GMT
           Session: 47112344
           Transport: RTP/AVP;unicast;client_port=3456-3457;
               server_port=6256-6257;mode="PLAY"



14.44 Unsupported

   The Unsupported response-header field lists the features not
   supported by the server. In the case where the feature was specified
   via the Proxy-Require field (Section 14.30), if there is a proxy on



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   the path between the client and the server, the proxy MUST send a
   response message with a status code of 551 (Option Not Supported).
   The request SHALL NOT be forwarded.

   See Section 14.35 for a usage example.

14.45 User-Agent

   See [H14.43] for explanation, however the syntax is clarified due to
   an error in RFC 2616. A Client SHOULD include this header in all RTSP
   messages it sends.

14.46 Vary

   See [H14.44]

14.47 Via

   See [H14.45].

14.48 WWW-Authenticate

   See [H14.47].

15 Caching

   In HTTP, response-request pairs are cached. RTSP differs
   significantly in that respect. Responses are not cacheable, with the
   exception of the presentation description returned by DESCRIBE.
   (Since the responses for anything but DESCRIBE and GET_PARAMETER do
   not return any data, caching is not really an issue for these
   requests.) However, it is desirable for the continuous media data,
   typically delivered out-of-band with respect to RTSP, to be cached,
   as well as the session description.

   On receiving a SETUP or PLAY request, a proxy ascertains whether it
   has an up-to-date copy of the continuous media content and its
   description. It can determine whether the copy is up-to-date by
   issuing a SETUP or DESCRIBE request, respectively, and comparing the
   Last-Modified header with that of the cached copy. If the copy is not
   up-to-date, it modifies the SETUP transport parameters as appropriate
   and forwards the request to the origin server. Subsequent control
   commands such as PLAY or PAUSE then pass the proxy unmodified. The
   proxy delivers the continuous media data to the client, while
   possibly making a local copy for later reuse. The exact behavior
   allowed to the cache is given by the cache-response directives
   described in Section 14.10. A cache MUST answer any DESCRIBE requests
   if it is currently serving the stream to the requestor, as it is



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   possible that low-level details of the stream description may have
   changed on the origin-server.

   Note that an RTSP cache, unlike the HTTP cache, is of the "cut-
   through" variety. Rather than retrieving the whole resource from the
   origin server, the cache simply copies the streaming data as it
   passes by on its way to the client. Thus, it does not introduce
   additional latency.

   To the client, an RTSP proxy cache appears like a regular media
   server, to the media origin server like a client. Just as an HTTP
   cache has to store the content type, content language, and so on for
   the objects it caches, a media cache has to store the presentation
   description.  Typically, a cache eliminates all transport-references
   (that is, multicast information) from the presentation description,
   since these are independent of the data delivery from the cache to
   the client.  Information on the encodings remains the same. If the
   cache is able to translate the cached media data, it would create a
   new presentation description with all the encoding possibilities it
   can offer.

16 Examples

   This section contains several different examples trying to illustrate
   possible ways of using RTSP. The examples can also help with the
   understanding of how functions of RTSP work. However remember that
   this is examples and the normative and syntax description in the
   other chapters takes precedence. Please also note that many of the
   example MAY contain syntax illegal line breaks to accommodate the
   formatting restriction that the RFC series impose.

16.1 Media on Demand (Unicast)

   Client C requests a movie distributed from two different media
   servers A (audio.example.com ) and V (video.example.com ). The media
   description is stored on a web server W. The media description
   contains descriptions of the presentation and all its streams,
   including the codecs that are available, dynamic RTP payload types,
   the protocol stack, and content information such as language or
   copyright restrictions. It may also give an indication about the
   timeline of the movie.

   In this example, the client is only interested in the last part of
   the movie.


   C->W: GET /twister.sdp HTTP/1.1
         Host: www.example.com



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         Accept: application/sdp

   W->C: HTTP/1.0 200 OK
         Date: 23 Jan 1997 15:35:06 GMT
         Content-Type: application/sdp
         Content-Length: 255
         Expires: 23 Jan 1998 15:35:06 GMT

         v=0
         o=- 2890844526 2890842807 IN IP4 192.16.24.202
         s=RTSP Session
         e=adm@example.com
         a=range:npt=0-1:49:34
         t=0 0
         m=audio 0 RTP/AVP 0
         a=control:rtsp://audio.example.com/twister/audio.en
         m=video 0 RTP/AVP 31
         a=control:rtsp://video.example.com/twister/video

   C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.0
         CSeq: 1
         User-Agent: PhonyClient/1.2
         Transport: RTP/AVP/UDP;unicast;client_port=3056-3057,
                    RTP/AVP/TCP;unicast;interleaved=0-1

   A->C: RTSP/1.0 200 OK
         CSeq: 1
         Session: 12345678
         Transport: RTP/AVP/UDP;unicast;client_port=3056-3057;
                    server_port=5000-5001
         Date: 23 Jan 1997 15:35:12 GMT
         Server: PhonyServer/1.0
         Expires: 24 Jan 1997 15:35:12 GMT
         Cache-Control: public
         Accept-Ranges: NPT, SMPTE

   C->V: SETUP rtsp://video.example.com/twister/video RTSP/1.0
         CSeq: 1
         User-Agent: PhonyClient/1.2
         Transport: RTP/AVP/UDP;unicast;client_port=3058-3059,
                    RTP/AVP/TCP;unicast;interleaved=0-1

   V->C: RTSP/1.0 200 OK
         CSeq: 1
         Session: 23456789
         Transport: RTP/AVP/UDP;unicast;client_port=3058-3059;
                    server_port=5002-5003
         Date: 23 Jan 1997 15:35:12 GMT



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         Server: PhonyServer/1.0
         Cache-Control: public
         Expires: 24 Jan 1997 15:35:12 GMT
         Accept-Ranges: NPT, SMPTE

   C->V: PLAY rtsp://video.example.com/twister/video RTSP/1.0
         CSeq: 2
         User-Agent: PhonyClient/1.2
         Session: 23456789
         Range: smpte=0:10:00-

   V->C: RTSP/1.0 200 OK
         CSeq: 2
         Session: 23456789
         Range: smpte=0:10:00-1:49:23
         RTP-Info: url=rtsp://video.example.com/twister/video;
                   seq=12312232;rtptime=78712811
         Server: PhonyServer/2.0
         Date: 23 Jan 1997 15:35:13 GMT

   C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/1.0
         CSeq: 2
         User-Agent: PhonyClient/1.2
         Session: 12345678
         Range: smpte=0:10:00-

   A->C: RTSP/1.0 200 OK
         CSeq: 2
         Session: 12345678
         Range: smpte=0:10:00-1:49:23
         RTP-Info: url=rtsp://audio.example.com/twister/audio.en;
                   seq=876655;rtptime=1032181
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:35:13 GMT



   C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/1.0
         CSeq: 3
         User-Agent: PhonyClient/1.2
         Session: 12345678

   A->C: RTSP/1.0 200 OK
         CSeq: 3
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:36:52 GMT

   C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/1.0



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         CSeq: 3
         User-Agent: PhonyClient/1.2
         Session: 23456789

   V->C: RTSP/1.0 200 OK
         CSeq: 3
         Server: PhonyServer/2.0
         Date: 23 Jan 1997 15:36:52 GMT




   Even though the audio and video track are on two different servers,
   and may start at slightly different times and may drift with respect
   to each other, the client can synchronize the two using standard RTP
   methods, in particular the time scale contained in the RTCP sender
   reports. Initial synchronization is achieved through the RTP-Info and
   Range headers information in the PLAY response.

16.2 Streaming of a Container file

   For purposes of this example, a container file is a storage entity in
   which multiple continuous media types pertaining to the same end-user
   presentation are present. In effect, the container file represents an
   RTSP presentation, with each of its components being RTSP streams.
   Container files are a widely used means to store such presentations.
   While the components are transported as independent streams, it is
   desirable to maintain a common context for those streams at the
   server end.


        This enables the server to keep a single storage handle
        open easily. It also allows treating all the streams
        equally in case of any prioritization of streams by the
        server.

   It is also possible that the presentation author may wish to prevent
   selective retrieval of the streams by the client in order to preserve
   the artistic effect of the combined media presentation. Similarly, in
   such a tightly bound presentation, it is desirable to be able to
   control all the streams via a single control message using an
   aggregate URL.

   The following is an example of using a single RTSP session to control
   multiple streams. It also illustrates the use of aggregate URLs. In a
   container file it is also desirable to not write any URL parts which
   is not kept, when the container is distributed, like the host and
   most of the path element. Therefore this example also uses the "*"



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   and relative URL in the delivered SDP.

   Client C requests a presentation from media server M. The movie is
   stored in a container file. The client has obtained an RTSP URL to
   the container file.


   C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/1.0
         CSeq: 1
         User-Agent: PhonyClient/1.2

   M->C: RTSP/1.0 200 OK
         CSeq: 1
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:35:06 GMT
         Content-Type: application/sdp
         Content-Length: 257
         Content-Base: rtsp://example.com/twister.3gp/
         Expires: 24 Jan 1997 15:35:06 GMT

         v=0
         o=- 2890844256 2890842807 IN IP4 172.16.2.93
         s=RTSP Session
         i=An Example of RTSP Session Usage
         e=adm@example.com
         a=control: *
         a=range: npt=0-0:10:34.10
         t=0 0
         m=audio 0 RTP/AVP 0
         a=control: trackID=1
         m=video 0 RTP/AVP 26
         a=control: trackID=4

   C->M: SETUP rtsp://example.com/twister.3gp/trackID=1 RTSP/1.0
         CSeq: 2
         User-Agent: PhonyClient/1.2
         Require: play.basic
         Transport: RTP/AVP;unicast;dest_addr=":8000"/":8001"

   M->C: RTSP/1.0 200 OK
         CSeq: 2
         Server: PhonyServer/1.0
         Transport: RTP/AVP;unicast;dest_addr=":8000"/":8001;
                    src_addr="172.16.2.93:9000"/"172.16.2.93:9001"
                    ssrc=93CB001E
         Session: 12345678
         Expires: 24 Jan 1997 15:35:12 GMT
         Date: 23 Jan 1997 15:35:12 GMT



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         Accept-Ranges: NPT

   C->M: SETUP rtsp://example.com/twister.3gp/trackID=4 RTSP/1.0
         CSeq: 3
         User-Agent: PhonyClient/1.2
         Require: play.basic
         Transport: RTP/AVP;unicast;dest_addr=":8002"/":8003"
         Session: 12345678

   M->C: RTSP/1.0 200 OK
         CSeq: 3
         Server: PhonyServer/1.0
         Transport: RTP/AVP;unicast;dest_addr=":8002"/":8003;
                    src_addr="172.16.2.93:9002"/"172.16.2.93:9003";
                    ssrc=A813FC13
         Session: 12345678
         Expires: 24 Jan 1997 15:35:13 GMT
         Date: 23 Jan 1997 15:35:13 GMT
         Accept-Range: NPT

   C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/1.0
         CSeq: 4
         User-Agent: PhonyClient/1.2
         Range: npt=0-10, npt=30-
         Session: 12345678

   M->C: RTSP/1.0 200 OK
         CSeq: 4
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:35:14 GMT
         Session: 12345678
         Range: npt=0-10, npt=30-623.10
         RTP-Info: url=rtsp://example.com/twister.3gp/trackID=4;
            seq=12345;rtptime=3450012,
           url=rtsp://example.com/twister.3gp/trackID=1;
            seq=54321;rtptime=2876889

   C->M: PAUSE rtsp://example.com/twister.3gp/ RTSP/1.0
         CSeq: 5
         User-Agent: PhonyClient/1.2
         Session: 12345678

   M->C: RTSP/1.0 200 OK
         CSeq: 5
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:36:01 GMT
         Session: 12345678
         Range: npt=34.57-623.10



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   C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/1.0
         CSeq: 6
         User-Agent: PhonyClient/1.2
         Range: npt=34.57-623.10
         Session: 12345678

   M->C: RTSP/1.0 200 OK
         CSeq: 6
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:36:01 GMT
         Session: 12345678
         Range: npt=34.57-623.10
         RTP-Info: url=rtsp://example.com/twister.3gp/trackID=4;
            seq=12555;rtptime=6330012,
           url=rtsp://example.com/twister.3gp/trackID=1;
            seq=55021;rtptime=3132889




16.3 Single Stream Container Files

   Some RTSP servers may treat all files as though they are "container
   files", yet other servers may not support such a concept. Because of
   this, clients SHOULD use the rules set forth in the session
   description for request URLs, rather than assuming that a consistent
   URL may always be used throughout. Here's an example of how a multi-
   stream server might expect a single-stream file to be served:


   C->S: DESCRIBE rtsp://foo.com/test.wav RTSP/1.0                        |
         Accept: application/x-rtsp-mh, application/sdp                   |
         CSeq: 1                                                          |
         User-Agent: PhonyClient/1.2                                      |

   S->C: RTSP/1.0 200 OK                                                  |
         CSeq: 1                                                          |
         Content-base: rtsp://foo.com/test.wav/                           |
         Content-type: application/sdp                                    |
         Content-length: 48                                               |
         Server: PhonyServer/1.0                                          |
         Date: 23 Jan 1997 15:35:06 GMT                                   |
         Expires: 23 Jan 1997 17:00:00 GMT                                |

         v=0                                                              |
         o=- 872653257 872653257 IN IP4 172.16.2.187                      |
         s=mu-law wave file                                               |
         i=audio test                                                     |



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         t=0 0                                                            |
         a=control: *                                                     |
         m=audio 0 RTP/AVP 0                                              |
         a=control:streamid=0                                             |

   C->S: SETUP rtsp://foo.com/test.wav/streamid=0 RTSP/1.0                |
         Transport: RTP/AVP/UDP;unicast;                                  |
            client_port=6970-6971;mode="PLAY"                             |
         CSeq: 2                                                          |
         User-Agent: PhonyClient/1.2                                      |

   S->C: RTSP/1.0 200 OK                                                  |
         Transport: RTP/AVP/UDP;unicast;client_port=6970-6971;            |
                     server_port=6970-6971;mode="PLAY";ssrc=EAB98712      |
         CSeq: 2                                                          |
         Session: 2034820394                                              |
         Expires: 23 Jan 1997 16:00:00 GMT                                |
         Server: PhonyServer/1.0                                          |
         Date: 23 Jan 1997 15:35:07 GMT                                   |

   C->S: PLAY rtsp://foo.com/test.wav/ RTSP/1.0                           |
         CSeq: 3                                                          |
         User-Agent: PhonyClient/1.2                                      |
         Session: 2034820394                                              |

   S->C: RTSP/1.0 200 OK                                                  |
         CSeq: 3                                                          |
         Server: PhonyServer/1.0                                          |
         Date: 23 Jan 1997 15:35:08 GMT                                   |
         Session: 2034820394                                              |
         Range: npt=0-600                                                 |
         RTP-Info: url=rtsp://foo.com/test.wav/streamid=0;                |
            seq=981888;rtptime=3781123                                    |



   Note the different URL in the SETUP command, and then the switch back
   to the aggregate URL in the PLAY command.  This makes complete sense
   when there are multiple streams with aggregate control, but is less
   than intuitive in the special case where the number of streams is
   one. However the server has declared that the aggregated control URL
   in the SDP and therefore this is fine.

   If however the server had not declared an aggregated control URL it
   would be another question, in which the client should consider it
   lucky if it works.

   In this case, it is also required that servers accept implementations



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   that uses the non-aggregated interpretation and uses the individual
   media URL, like this:


   C->S: PLAY rtsp://example.com/test.wav/streamid=0 RTSP/1.0
         CSeq: 3
         User-Agent: PhonyClient/1.2



16.4 Live Media Presentation Using Multicast

   The media server M chooses the multicast address and port. Here, we
   assume that the web server only contains a pointer to the full
   description, while the media server M maintains the full description.

   Editors note: Is this example really valid? In what situations does
   it make sense to do a setup to a multicast distribution channel, and
   also issue PLAY requests?


   C->W: GET /sessions.html HTTP/1.1
         Host: www.example.com

   W->C: HTTP/1.1 200 OK
         Content-Type: text/html

         <html>
           ...
           <href "Stremed Live Music performance"
              src="rtsp://live.example.com/concert/audio">
           ...
         </html>

   C->M: DESCRIBE rtsp://live.example.com/concert/audio RTSP/1.0
         CSeq: 1
         Supported: play.basic, play.scale

   M->C: RTSP/1.0 200 OK
         CSeq: 1
         Content-Type: application/sdp
         Content-Length: 181
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:35:06 GMT
         Supported: play.basic

         v=0
         o=- 2890844526 2890842807 IN IP4 192.16.24.202



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         s=RTSP Session
         m=audio 3456 RTP/AVP 0
         c=IN IP4 224.2.0.1/16
         a=control: rtsp://live.example.com/concert/audio
         a=range:npt=0-

   C->M: SETUP rtsp://live.example.com/concert/audio RTSP/1.0
         CSeq: 2
         Transport: RTP/AVP;multicast

   M->C: RTSP/1.0 200 OK
         CSeq: 2
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:35:06 GMT
         Transport: RTP/AVP;multicast;destination=224.2.0.1;
                    port=3456-3457;ttl=16
         Session: 0456804596
         Accept-Ranges: NPT, UTC

   C->M: PLAY rtsp://live.example.com/concert/audio RTSP/1.0
         CSeq: 3
         Session: 0456804596

   M->C: RTSP/1.0 200 OK
         CSeq: 3
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:35:07 GMT
         Session: 0456804596
         Range:npt=1256-
         RTP-Info: url=rtsp://live.example.com/concert/audio;
                   seq=1473; rtptime=80000



16.5 Capability Negotiation

   This examples illustrate how the client and server determines there
   capability to support a special feature, in this case "play.scale".
   The server through the clients request, and included Supported header
   learns that the client is supporting this updated specification, and
   also support the playback time scaling feature of RTSP. The server's
   response declares that it is also an updated specification minimal
   implementation and supports the extra features, of client requested
   time scaling and faster than normal transmission rates, plus one
   "example.com" proprietary feature "flight". The client also learns
   what methods that are possible to use in regards to the indicated
   resource.




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   C->S: OPTIONS rtsp://media.example.com/movie/twister.3gp RTSP/1.0
         CSeq: 1
         Supported: play.basic, play.scale
         User-Agent: PhonyClient/1.2

   S->C: RTSP/1.0 200 OK
         CSeq: 1
         Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN
         Server: PhonyServer/2.0
         Supported: play.basic, play.scale, example.com.flight



   When the client sends its SETUP request it tells the server that it
   is must support the play.scale feature for this session by including
   the Require header.


   C->S: SETUP rtsp://media.example.com/twister.3gp/trackID=1 RTSP/1.0
         CSeq: 3
         User-Agent: PhonyClient/1.2
         Transport: RTP/AVP/UDP;unicast;client_port=3056-3057,
                    RTP/AVP/TCP;unicast;interleaved=0-1
         Require: play.scale

   S->C: RTSP/1.0 200 OK
         CSeq: 3
         Session: 12345678
         Transport: RTP/AVP/UDP;unicast;client_port=3056-3057;
                    server_port=5000-5001
         Server: PhonyServer/2.0
         Accept-Ranges: NPT, SMPTE



17 Security Framework

   The RTSP security framework consists of two high level components:
   the pure authentication mechanisms based on HTTP authentication, and
   the transport protection based on TLS, which is independent of RTSP.
   Because of the similarity in syntax and usage between RTSP servers
   and HTTP servers, the security for HTTP is re-used to a large extent.

17.1 RTSP and HTTP Authentication


   RTSP and HTTP share common authentication schemes, and thus follow     |
   the same usage guidelines as specified in  [8] and also in [H15].      |



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   Servers SHOULD implement both basic and digest [8] authentication.     |

   It should be stressed that using the HTTP authentication alone does    |
   not provide full control message security. Therefore, in environments  |
   requiring tighter security for the control messages, TLS SHOULD be     |
   used, see Section  17.2.                                               |

17.2 RTSP over TLS                                                        |

   RTSP SHALL follow the same guidelines with regards to TLS [7] usage    |
   as specified for HTTP, see [19]. RTSP over TLS is separated from       |
   unsecured RTSP both on URI level and port level. Instead of using the  |
   "rtsp" scheme identifier in the URI, the "rtsps" scheme identifier     |
   MUST be used to signal RTSP over TLS. If no port is given in a URI     |
   with the "rtsps" scheme, port 322 SHALL be used for TLS over TCP/IP.   |

   When a client tries to setup an insecure channel to the server (using  |
   the "rtsp" URI), and the policy for the resource requires a secure     |
   channel, the server SHALL redirect the client to the secure service    |
   by sending a 301 redirect response code together with the correct      |
   Location URI (using the "rtsps" scheme).                               |

   It should be noted that TLS allows for mutual authentication (when     |
   using both server and client certificates). Still, one of the more     |
   common way TLS is used is to only provide server side authentication   |
   (often to avoid client certificates). TLS is then used in addition to  |
   HTTP authentication, providing transport security and server           |
   authentication, while HTTP Authentication is used to authenticate the  |
   client.                                                                |

   RTSP includes the possibility to keep a TCP session up between the     |
   client and server, throughout the RTSP session lifetime. It may be     |
   convenient to keep the TCP session, not only to save the extra setup   |
   time for TCP, but also the extra setup time for TLS (even if TLS uses  |
   the resume function, there will be almost two extra roundtrips).       |
   Still, when TLS is used, such behavior introduces extra active state   |
   in the server, not only for TCP and RTSP, but also for TLS. This may   |
   increase the vulnerability to DoS attacks.                             |

   In addition to these recommendations, Section 17.3 gives further       |
   recommendations of TLS usage with proxies.                             |

17.3 Security and Proxies                                                 |

   The nature of a proxy is often to act as a "man-in-the-middle", while  |
   security is often about preventing the existence of a "man-in-the-     |
   middle".                                                               |




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   There are basically two categories of inspecting proxies, the          |
   transparent proxies (which the client is not aware of) and the non-    |
   transparent proxies (which the client is aware of). An infrastructure  |
   based on proxies must assume that the trust model is such that both    |
   client and servers can trust the proxies to handle the RTSP messages   |
   correctly. To be able to trust a proxy, the client and server must     |
   also be aware of the proxy. Hence, transparent proxies cannot          |
   generally be seen as trusted and will not work well with security      |
   (unless they work only at transport layer and not above that in the    |
   stack). In the rest of this section we will by proxy refer to a non-   |
   transparent proxy, which requires to inspect/manipulate the RTSP       |
   messages.                                                              |

   The HTTP Authentication is built on the assumption of proxies and can  |
   provide user-proxy authentication and proxy-proxy/server               |
   authentication in addition to the client-server authentication.        |

   When TLS is applied and a proxy is used, the client will use the       |
   proxy's destination URI address when sending messages. This implies    |
   that for TLS, the client will authenticate the proxy server and not    |
   the end server. Note that, when the client checks the server           |
   certificate in TLS, it MUST check the proxy's identity (URI or         |
   possibly other known identity) against the proxy's identity as         |
   presented in the proxy's Certificate message.                          |

   A problem here is that, even though the client accepts the proxy, the  |
   proxy has no information on what grounds it shall accept the           |
   server/next proxy certificate (it may of course have own decision      |
   rules for this, but very likely the user may have different as well).  |
   To handle this case, the Accept-Credentials header (See Section 14.2)  |
   is used, where the client can force the proxy/proxies to relay back    |
   the certificates used by any intermediate proxies as well as the       |
   server. Given the assumption that the proxies are view as trusted, it  |
   gives the user a possibility to enforce policies to each trusted       |
   proxy of whether it should accept the next entity in the chain.        |

   A proxy MUST use TLS for the next hop if the RTSP request includes a   |
   "rtsps" URI. TLS MAY be applied on intermediate links (e.g. between    |
   client and proxy, or between proxy and proxy), even if the resource    |
   and the end server does not require to use it.                         |

17.3.1 Accept-Credentials                                                 |

   The Accept-Credentials header can be used by the client to distribute  |
   simple authorization policies to intermediate proxies. The client      |
   includes the Accept-Credentials header to dictate how the proxy shall  |
   treat the server/next proxy certificate. There are currently three     |
   methods defined:                                                       |



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        Any, which means that the proxy (or proxies) SHALL accept         |
             whatever certificate presented. This is of course not a      |
             recommended option to use, but may be useful in certain      |
             circumstances (such as testing).                             |

        Proxy, which means that the proxy (or proxies) MUST use its own   |
             policies to validate the certificate and decided whether to  |
             accept it or not. This may be convenient in cases where it   |
             is a personal proxy or the client in some other way have a   |
             very strong trust relation with the proxy (e.g. the proxy    |
             includes other out-of-band possibilities for the client to   |
             configure policies in the proxy).                            |

        User, which means that the proxy (or proxies) MUST send           |
             credential information about the next hop to the client for  |
             authorization. The client can then decide whether the proxy  |
             should accept the certificate or not. See section 17.3.2     |
             for further details.                                         |

   If the Accept-Credentials header is not included in the RTSP request   |
   from the client, the default method used SHALL be "Proxy". If          |
   something else than the "Proxy" method is used, the Accept-            |
   Credentials header SHALL always be included in the RTSP request from   |
   the client. This is because it cannot be assumed that the proxy        |
   always keeps the TLS state or the users previously preference between  |
   different RTSP messages (in particular if the time interval between    |
   the messages is long).                                                 |

   The "Any" and "Proxy" methods does not require the proxy to provide    |
   any specific response, but only apply the policy as defined for        |
   respectively method. If the policy does not accept the credentials of  |
   the next hop, the entity SHALL respond with a message using status     |
   code 471 (Connection Credentials not accepted).                        |

   An RTSP request in the direction server to client MUST NOT include     |
   the Accept-Credential header. As for the non-secured communication,    |
   the possibility for these request depend on the presence of a client   |
   established connection. However if the server to client request is in  |
   relation to a session established over a TLS secured channel, if MUST  |
   be sent in a TLS secured connection. That secured connection MUST      |
   also be the one used by the last client to server request. If no such  |
   transport connection exist at the time when the server desire to send  |
   the request, it silently fails.                                        |

   Further policies MAY be defined and registered, but should be done so  |
   with caution.                                                          |

17.3.2 User approved TLS procedure                                        |



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   For the "User" method each proxy MUST perform the the following        |
   procedure for each RTSP request:                                       |

        o Setup the TLS session to the next hop if not already present    |
          (i.e. run the TLS handshake, but do not send the RTSP           |
          request).                                                       |

        o Extract the peer certificate for the TLS session.               |

        o Check if a matching identity and hash of the peer certificate   |
          is present in the Accept-Credentials header.  If present, send  |
          the message to the next hop, and conclude these procedures. If  |
          not, go to the next step.                                       |

        o The proxy responds to the RTSP request with a 470 or 407        |
          response code. The 407 response code MAY be used when the       |
          proxy requires both user and connection authorization from      |
          user or client. In this message the proxy SHALL include a       |
          Accept-Credentials header with the next hop's identity and      |
          certificate.                                                    |

   The client MUST upon receiving a 470 or 407 response with Accept-      |
   Credentials take the decision on whether to accept the certificate or  |
   not (if it cannot do so, the user SHOULD be consulted). If the         |
   certificate is accepted, the client have to again send the RTSP        |
   request. In that request the client have to include the Accept-        |
   Credentials header including the hash over the DER encoded             |
   certificate for all trusted proxies in the chain.                      |



   Example:                                                               |
     Accept-Credentials:User,                                             |
          "rtsps://proxy2.example.com/";exaIl9VMbQMOFGClx5rXnPJKVNI=,     |
          "rtsps://server.example.com/";lurbjj5khhB0NhIuOXtt4bBRH1M=      |




   The implication is that the connection for secured RTSP messages may   |
   take significantly more round-trip times for the first message. As     |
   each proxy that has to be accepted will result in a complete extra     |
   message exchange between the proxy connecting to the next hop and the  |
   client. However after the first message exchange the remaining         |
   message should not be delayed, as each message shall contain the       |
   chain of proxies that the requestor accepts. The procedure of          |
   including the credentials in each request rather than building state   |
   in each proxy, avoids the need for revocation procedures.              |



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18 Syntax

   The RTSP syntax is described in an augmented Backus-Naur Form (BNF)
   as defined in RFC 2234 [5].

18.1 Base Syntax


   OCTET      =  %x00-FF <any 8-bit sequence of data>
   CHAR       =  %x01-7F <any US-ASCII character (octets 1 - 127)>
   UPALPHA    =  %x41-5A <any US-ASCII uppercase letter "A".."Z">
   LOALPHA    =  %x61-7A <any US-ASCII lowercase letter "a".."z">
   ALPHA      =  UPALPHA / LOALPHA
   DIGIT      =  %x30-39 <any US-ASCII digit "0".."9">
   CTL        =  %x00-1F / %x7F <any US-ASCII control character
                 (octets 0 - 31) and DEL (127)>
   CR         =  %x0D <US-ASCII CR, carriage return (13)>
   LF         =  %x0A <US-ASCII LF, linefeed (10)>
   SP         =  %x20 <US-ASCII SP, space (32)>
   HT         =  %x09 <US-ASCII HT, horizontal-tab (9)>
   DQUOTE     =  %x22 <US-ASCII double-quote mark (34)>
   BACKSLASH  =  %x5C <US-ASCII backslash (92)>
   CRLF       =  CR LF



   LWS            =  [CRLF] 1*( SP / HT )
   TEXT           =  %x20-7D / %x80-FF <any OCTET except CTLs>
   tspecials      =  "(" / ")" / "<" / ">" / "@"
                  /  "," / ";" / ":" / BACKSLASH / DQUOTE
                  /  "/" / "[" / "]" / "?" / "="
                  /  "{" / "}" / SP / HT
   token          =  1*(%x21 / %x23-27 / %x2A-2B / %x2D-2E / %x30-39
                  /  %x41-5A / %x5E-7A / %x7C / %x7E)
                     1*<any CHAR except CTLs or tspecials>
   quoted-string  =  ( DQUOTE *(qdtext) DQUOTE )
   qdtext         =  %x20-21 / %x23-7D / %x80-FF <any TEXT except <">>
   quoted-pair    =  BACKSLASH CHAR



   safe         =  "$" / "-" / "_" / "." / "+"
   extra        =  "!" / "*" / "'" / "(" / ")" / ","
   hex          =  DIGIT / "A" / "B" / "C" / "D" / "E" / "F" /
                   "a" / "b" / "c" / "d" / "e" / "f"
   escape       =  "%" hex hex
   reserved     =  ";" / "/" / "?" / ":" / "@" / "&" / "="
   unreserved   =  alpha / digit / safe / extra



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   xchar        =  unreserved / reserved / escape
   base64       =  0*base64-unit [base64-pad]
   base64-unit  =  4base64-char
   base64-pad   =  (2base64-char "==") / (3base64-char "=")
   base64-char  =  ALPHA / DIGIT / "+" / "/"


18.2 RTSP Protocol Definition

18.2.1 Generic Protocol elements


   absoluteURL  =  as defined in RFC 2396  [13] and RFC2732  [12]
   relativeURL  =  as defined in RFC 2396  [13] and RFC2732  [12]
   rtsp_URL     =  rtsp-scheme "//" host [":" port]
                   [abs_path ["?" query]] ["#" fragment]
   rtsp-scheme  = ( "rtsp:" / "rtspu:" / "rtsps:" )
   host         =  As defined by RFC 2732 [12]
   abs_path     =  As defined by RFC 2396 [13]
   port         =  *DIGIT ; Is expected to be 1*5DIGIT
   query        =  As defined by RFC 2396 [13]
   fragment     =  As defined by RFC 2396 [13]



   smpte-range           =  smpte-type "=" smpte-range-spec
                            ;Section 3.4
   smpte-range-spec      =  ( smpte-time "-" [ smpte-time ] )
                         /  ( "-" smpte-time )
   smpte-type            =  "smpte" / "smpte-30-drop"
                         /  "smpte-25" / smpte-type-extension
                            ; other timecodes may be added
   smpte-type-extension  =  token
   smpte-time            =  1*2DIGIT ":" 1*2DIGIT ":" 1*2DIGIT
                            [ ":" 1*2DIGIT [ "." 1*2DIGIT ] ]



   npt-range       =  ["npt" "="] npt-range-spec ; Section 3.5
                      ; implementations SHOULD use npt= prefix,
                      ;but SHOULD be prepared to interoperate with
                      ; RFC 2326 implementations which don't use it.
   npt-range-spec  =  ( npt-time "-" [ npt-time ] ) / ( "-" npt-time )
   npt-time        =  "now" / npt-sec / npt-hhmmss
   npt-sec         =  1*DIGIT [ "." *DIGIT ]
   npt-hhmmss      =  npt-hh ":" npt-mm ":" npt-ss [ "." *DIGIT ]
   npt-hh          =  1*DIGIT ; any positive number
   npt-mm          =  1*2DIGIT ; 0-59



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   npt-ss          =  1*2DIGIT ; 0-59



   utc-range       =  "clock" "=" utc-range-spec ; Section 3.6
   utc-range-spec  =  ( utc-time "-" [ utc-time ] ) / ( "-" utc-time )
   utc-time        =  utc-date "T" utc-clock "Z"
   utc-date        =  8DIGIT ; < YYYYMMDD >
   utc-clock       =  6DIGIT [ "." fraction ]; < HHMMSS.fraction >
   fraction        =  1*DIGIT



   feature-tag     =  token
   session-id      =  8*( ALPHA / DIGIT / safe )
   message-header  =  field-name ":" [ field-value ] CRLF
   field-name      =  token
   field-value     =  *( field-content / LWS )
   field-content   =  <the OCTETs making up the field-value and
                      consisting of either *TEXT or combinations
                      of token, tspecials, and quoted-string>


18.2.2 Message Syntax



        RTSP-message  =   Request / Response ; RTSP/1.0 messages
        Request       =   Request-Line        ; Section 6.1
                      *(  general-header      ; Section 5
                      /   request-header      ; Section 6.2
                      /   entity-header )     ; Section 8.1
                          CRLF
                          [ message-body ]    ; Section 4.3
        Response      =   Status-Line         ; Section 7.1
                      *(  general-header      ; Section 5
                      /   response-header     ; Section 7.1.2
                      /   entity-header )     ; Section 8.1
                          CRLF
                          [ message-body ]    ; Section 4.3



   Request-Line  =  Method SP Request-URI SP RTSP-Version CRLF
   Status-Line   =  RTSP-Version SP Status-Code SP Reason-Phrase CRLF






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   Method            =  "DESCRIBE"        ; Section 11.2
                     /  "GET_PARAMETER"   ; Section 11.7
                     /  "OPTIONS"         ; Section 11.1
                     /  "PAUSE"           ; Section 11.5
                     /  "PLAY"            ; Section 11.4
                     /  "PING"            ; Section 11.10
                     /  "REDIRECT"        ; Section 11.9
                     /  "SETUP"           ; Section 11.3
                     /  "SET_PARAMETER"   ; Section 11.8
                     /  "TEARDOWN"        ; Section 11.6
                     /  extension-method
   extension-method  =  token



   Request-URI   =  "*" / absolute_URL
   RTSP-Version  =  "RTSP" "/" 1*DIGIT "." 1*DIGIT




        Status-Code  =  "100" ; Continue
                     /  "200" ; OK
                     /  "201" ; Created
                     /  "250" ; Low on Storage Space
                     /  "300" ; Multiple Choices
                     /  "301" ; Moved Permanently
                     /  "302" ; Moved Temporarily
                     /  "303" ; See Other
                     /  "304" ; Not Modified
                     /  "305" ; Use Proxy
                     /  "400" ; Bad Request
                     /  "401" ; Unauthorized
                     /  "402" ; Payment Required
                     /  "403" ; Forbidden
                     /  "404" ; Not Found
                     /  "405" ; Method Not Allowed
                     /  "406" ; Not Acceptable
                     /  "407" ; Proxy Authentication Required
                     /  "408" ; Request Time-out
                     /  "410" ; Gone
                     /  "411" ; Length Required
                     /  "412" ; Precondition Failed
                     /  "413" ; Request Entity Too Large
                     /  "414" ; Request-URI Too Large
                     /  "415" ; Unsupported Media Type
                     /  "451" ; Parameter Not Understood
                     /  "452" ; reserved



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                     /  "453" ; Not Enough Bandwidth
                     /  "454" ; Session Not Found
                     /  "455" ; Method Not Valid in This State
                     /  "456" ; Header Field Not Valid for Resource
                     /  "457" ; Invalid Range
                     /  "458" ; Parameter Is Read-Only
                     /  "459" ; Aggregate operation not allowed
                     /  "460" ; Only aggregate operation allowed
                     /  "461" ; Unsupported transport
                     /  "462" ; Destination unreachable
                     /  "470" ; Connection Authorization Required
                     /  "471" ; Connection Credentials not accepted
                     /  "500" ; Internal Server Error
                     /  "501" ; Not Implemented
                     /  "502" ; Bad Gateway
                     /  "503" ; Service Unavailable
                     /  "504" ; Gateway Time-out
                     /  "505" ; RTSP Version not supported
                     /  "551" ; Option not supported
                     /  extension-code



        extension-code  =  3DIGIT
        Reason-Phrase   =  *<TEXT, excluding CR, LF>



   general-header  =  Cache-Control     ; Section 14.10
                   /  Connection        ; Section 14.11
                   /  CSeq              ; Section 14.18
                   /  Date              ; Section 14.19
                   /  Supported         ; Section 14.41
                   /  Timestamp         ; Section 14.42
                   /  Via               ; Section 14.47
                   /  extension-header




   request-header  =  Accept              ; Section 14.1 and [H14.1]      |
                   /  Accept-Credentials  ; Section 14.2                  |
                   /  Accept-Encoding     ; Section 14.3 and [H14.3]      |
                   /  Accept-Language     ; Section 14.4 and [H14.4]      |
                   /  Authorization       ; Section 14.7 and [H14.8]      |
                   /  Bandwidth           ; Section 14.8                  |
                   /  Blocksize           ; Section 14.9                  |
                   /  From                ; Section 14.22                 |



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                   /  If-Match            ; Section 14.24                 |
                   /  If-Modified-Since   ; Section 14.25 and [H14.25]    |
                   /  If-None-Match       ; Section 14.26                 |
                   /  Proxy-Require       ; Section 14.30                 |
                   /  Range               ; Section 14.32                 |
                   /  Referer             ; Section 14.33                 |
                   /  Require             ; Section 14.35                 |
                   /  Scale               ; Section 14.37                 |
                   /  Session             ; Section 14.40                 |
                   /  Speed               ; Section 14.38                 |
                   /  Supported           ; Section 14.41                 |
                   /  Transport           ; Section 14.43                 |
                   /  User-Agent          ; Section 14.45                 |
                   /  extension-header                                    |



   response-header  =  Accept-Credentials  ; Section 14.2                 |
                    /  Accept-Ranges       ; Section 14.5                 |
                    /  ETag                ; Section 14.20                |
                    /  Location            ; Section 14.28                |
                    /  Proxy-Authenticate  ; Section 14.29                |
                    /  Public              ; Section 14.31                |
                    /  Range               ; Section 14.32                |
                    /  Retry-After         ; Section 14.34                |
                    /  RTP-Info            ; Section 14.36                |
                    /  Scale               ; Section 14.37                |
                    /  Session             ; Section 14.40                |
                    /  Server              ; Section 14.39                |
                    /  Speed               ; Section 14.38                |
                    /  Transport           ; Section 14.43                |
                    /  Unsupported         ; Section 14.44                |
                    /  Vary                ; Section 14.46                |
                    /  WWW-Authenticate    ; Section 14.48                |
                    /  extension-header                                   |



   entity-header     =  Allow             ; Section 14.6
                     /  Content-Base      ; Section 14.12
                     /  Content-Encoding  ; Section 14.13
                     /  Content-Language  ; Section 14.14
                     /  Content-Length    ; Section 14.15
                     /  Content-Location  ; Section 14.16
                     /  Content-Type      ; Section 14.17
                     /  Expires           ; Section 14.21 and [H14.21]
                     /  Last-Modified     ; Section 14.27
                     /  extension-header



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   extension-header  =  message-header


18.2.3 Header Syntax

   All header syntaxes not defined in this section are defined in
   chapter 14 of the HTTP 1.1 specification [4].                          |

   accept-credentials   =  "Accept-Credentials" ":" accept-cred-data      |
   accept-cred-data     =  credential-decision / credential-info          |
   credential-decision  =  ("User" "," [credential-info])                 |
                        /  "Proxy"                                        |
                        /  "Any"                                          |
                        /  token ; For future extensions                  |
   credential-info      =  cred-info-data 0*("," cred-info-data)          |
   cred-info-data       =  DQUOTE rtsp_URL DQUOTE ";" base64              |
   Accept-Ranges        =  "Accept-Ranges" ":" acceptable-ranges          |
   acceptable-ranges    =  (range-unit *("," LWS range-unit))             |
                        /  "none"                                         |
   range-unit           =  NPT / SMPTE / UTC / extension-format           |
   extension-format     =  token                                          |
   Bandwidth            =  "Bandwidth" ":" 1*DIGIT                        |
   Blocksize            =  "Blocksize" ":" 1*DIGIT                        |




   Cache-Control             =  "Cache-Control" ":" cache-directive
                                *("," LWS cache-directive)
   cache-directive           =  cache-request-directive
                             /  cache-response-directive
   cache-request-directive   =  "no-cache"
                             /  "max-stale" ["=" delta-seconds]
                             /  "min-fresh" "=" delta-seconds
                             /  "only-if-cached"
                             /  cache-extension
   cache-response-directive  =  "public"
                             /  "private"
                             /  "no-cache"
                             /  "no-transform"
                             /  "must-revalidate"
                             /  "proxy-revalidate"
                             /  "max-age" "=" delta-seconds
                             /  cache-extension
   cache-extension           =  token ["=" (token / quoted-string)]
   delta-seconds             =  1*DIGIT





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   Content-Base   =  "Content-Base" ":" absoluteURL
   CSeq           =  "Cseq" ":" 1*DIGIT
   Proxy-Require  =  "Proxy-Require" ":" feature-tag
                     *("," LWS feature-tag)
   Public         =  "Public" ":" method *("," LWS method)
   Range          =  "Range" ":" ranges-spec *("," LWS ranges-spec)
                     [ ";" "time" "=" utc-time ]
   ranges-spec    =  npt-range / utc-range / smpte-range
   Require        =  "Require" ":" feature-tag *("," LWS feature-tag)



   RTP-Info        =  "RTP-Info" ":" rtsp-info-spec
                      *("," LWS rtsp-info-spec)
   rtsp-info-spec  =  stream-url 1*parameter
   stream-url      =  quoted-url / unquoted-url
   unquoted-url    =  "url" "=" safe-url
   quoted-url      =  "url" "=" DQUOTE needquote-url DQUOTE
   safe-url        =  url
   needquote-url   =  url //That contains ; or ,
   url             =  ( absoluteURL / relativeURL )
   parameter       =  ";" "seq" "=" 1*DIGIT
                   /  ";" "rtptime" "=" 1*DIGIT



   Scale      =  "Scale" ":" [ "-" ] 1*DIGIT [ "." *DIGIT ]
   Speed      =  "Speed" ":" 1*DIGIT [ "." *DIGIT ]
   Server     =  "Server" ":" ( product / comment )
                 *(SP (product / comment))
   Session    =  "Session" ":" session-id
                 [ ";" "timeout" "=" delta-seconds ]
   Supported  =  "Supported" ":" [feature-tag *("," LWS feature-tag)]



   Timestamp       =  "Timestamp" ":" *(DIGIT) ["." *(DIGIT)] [delay]
   delay           =  *(DIGIT) [ "." *(DIGIT) ]
   Transport       =  "Transport" ":" transport-spec
                      *("," LWS transport-spec)
   transport-spec  =  transport-id *parameter
   transport-id    =  transport-prot "/" profile ["/" lower-transport]
                      ; no LWS is allowed inside transport-id



   transport-prot    =  "RTP" / token
   profile           =  "AVP" / token



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   lower-transport   =  "TCP" / "UDP" / token
   parameter         =  ";" ( "unicast" / "multicast" )
                    /   ";" "source" "=" host
                    /   ";" "destination" [ "=" host ]
                    /   ";" "interleaved" "=" channel [ "-" channel ]
                    /   ";" "append"
                    /   ";" "ttl" "=" ttl
                    /   ";" "layers" "=" 1*DIGIT
                    /   ";" "port" "=" port-spec
                    /   ";" "client_port" "=" port-spec
                    /   ";" "server_port" "=" port-spec
                    /   ";" "ssrc" "=" ssrc *("/" ssrc)
                    /   ";" "client_ssrc" "=" ssrc
                    /   ";" "mode" "=" mode-spec
                    /   ";" "dest_addr" "=" addr-list
                    /   ";" "src_addr" "=" addr-list
                    /   ";" trn-param-ext
   port-spec         =  port [ "-" port ]
   trn-param-ext     =  par-name "=" trn-par-value
   par-name          =  token
   trn-par-value     =  *(unreserved / DQUOTE *TEXT DQUOTE)
   ttl               =  1*3(DIGIT)
   ssrc              =  8*8(HEX)
   channel           =  1*3(DIGIT)
   mode-spec         =  ( DQUOTE mode *("," *SP mode) DQUOTE ) / mode
   mode              =  "PLAY" / "RECORD" / token
   addr-list         =  quoted-host-port *("/" quoted-host-port)
   quoted-host-port  =  DQUOTE host [":" port] DQUOTE



   Unsupported  =  "Unsupported" ":" feature-tag *("," feature-tag)
   User-Agent   =  "User-Agent" ":" ( product / comment )
                   0*(SP (product / comment)


19 Security Considerations

   Because of the similarity in syntax and usage between RTSP servers
   and HTTP servers, the security considerations outlined in [H15]
   apply.  Specifically, please note the following:

        Abuse of Server Log Information: RTSP and HTTP servers will       |
             presumably have similar logging mechanisms, and thus should
             be equally guarded in protecting the contents of those
             logs, thus protecting the privacy of the users of the
             servers. See [H15.1.1] for HTTP server recommendations
             regarding server logs.



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        Transfer of Sensitive Information: There is no reason to believe
             that information transferred via RTSP may be any less
             sensitive than that normally transmitted via HTTP.
             Therefore, all of the precautions regarding the protection
             of data privacy and user privacy apply to implementors of
             RTSP clients, servers, and proxies. See [H15.1.2] for
             further details.

        Attacks Based On File and Path Names: Though RTSP URLs are
             opaque handles that do not necessarily have file system
             semantics, it is anticipated that many implementations will
             translate portions of the request URLs directly to file
             system calls. In such cases, file systems SHOULD follow the
             precautions outlined in [H15.5], such as checking for ".."
             in path components.

        Personal Information: RTSP clients are often privy to the same
             information that HTTP clients are (user name, location,
             etc.)  and thus should be equally. See [H15.1] for further
             recommendations.

        Privacy Issues Connected to Accept Headers: Since may of the
             same "Accept" headers exist in RTSP as in HTTP, the same
             caveats outlined in [H15.1.4] with regards to their use
             should be followed.

        DNS Spoofing: Presumably, given the longer connection times
             typically associated to RTSP sessions relative to HTTP
             sessions, RTSP client DNS optimizations should be less
             prevalent.  Nonetheless, the recommendations provided in
             [H15.3] are still relevant to any implementation which
             attempts to rely on a DNS-to-IP mapping to hold beyond a
             single use of the mapping.

        Location Headers and Spoofing: If a single server supports
             multiple organizations that do not trust one another, then
             it must check the values of Location and Content-Location
             header fields in responses that are generated under control
             of said organizations to make sure that they do not attempt
             to invalidate resources over which they have no authority.
             ([H15.4])

   In addition to the recommendations in the current HTTP specification
   (RFC 2616 [4], as of this writing) and also of the previous RFC2068
   [20], future HTTP specifications may provide additional guidance on
   security issues.

   The following are added considerations for RTSP implementations.



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        Concentrated denial-of-service attack: The protocol offers the
             opportunity for a remote-controlled denial-of-service
             attack.

             The attacker may initiate traffic flows to one or more IP
             addresses by specifying them as the destination in SETUP
             requests. While the attacker's IP address may be known in
             this case, this is not always useful in prevention of more
             attacks or ascertaining the attackers identity. Thus, an
             RTSP server SHOULD only allow client-specified destinations
             for RTSP-initiated traffic flows if the server has verified
             the client's identity, either against a database of known
             users using RTSP authentication mechanisms (preferably
             digest authentication or stronger), or other secure means.

        Session hijacking: Since there is no or little relation between
             a transport layer connection and an RTSP session, it is
             possible for a malicious client to issue requests with
             random session identifiers which would affect unsuspecting
             clients. The server SHOULD use a large, random and non-
             sequential session identifier to minimize the possibility
             of this kind of attack.


        Authentication: Servers SHOULD implement both basic and digest    |
             [8] authentication. In environments requiring tighter        |
             security for the control messages, the transport layer       |
             mechanism TLS (RFC 2246 [7]) SHOULD be used.

        Stream issues: RTSP only provides for stream control. Stream
             delivery issues are not covered in this section, nor in the
             rest of this draft. RTSP implementations will most likely
             rely on other protocols such as RTP, IP multicast, RSVP and
             IGMP, and should address security considerations brought up
             in those and other applicable specifications.

        Persistently suspicious behavior: RTSP servers SHOULD return
             error code 403 (Forbidden) upon receiving a single instance
             of behavior which is deemed a security risk. RTSP servers
             SHOULD also be aware of attempts to probe the server for
             weaknesses and entry points and MAY arbitrarily disconnect
             and ignore further requests clients which are deemed to be
             in violation of local security policy.

20 IANA Considerations

   This section set up a number of registers for RTSP that should be
   maintained by IANA. For each registry there is a description on what



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   it shall contain, what specification is needed when adding a entry
   with IANA, and finally the entries that this document needs to
   register. See also the section 1.6 "Extending RTSP". There is also an
   IANA registration of two SDP attributes.

   The sections describing how to register an item uses some of the
   requirements level described in RFC 2434 [21], namely " First Come,
   First Served", "Specification Required", and "Standards Action".

   A registration request to IANA MUST contain the following
   information:

        o A name of the item to register according to the rules
          specified by the intended registry.

        o Indication of who has change control over the feature (for
          example, IETF, ISO, ITU-T, other international standardization
          bodies, a consortium, a particular company or group of
          companies, or an individual);

        o A reference to a further description, if available, for
          example (in order of preference) an RFC, a published standard,
          a published paper, a patent filing, a technical report,
          documented source code or a computer manual;

        o For proprietary features, contact information (postal and
          email address);

20.1 Feature-tags

20.1.1 Description

   When a client and server try to determine what part and functionality
   of the RTSP specification and any future extensions that its counter
   part implements there is need for a namespace.  This registry
   contains named entries representing certain functionality.

   The usage of feature-tags is explained in section 10 and 11.1.

20.1.2 Registering New Feature-tags with IANA

   The registering of feature-tags is done on a first come, first served
   basis.

   The name of the feature MUST follow these rules: The name may be of
   any length, but SHOULD be no more than twenty characters long. The
   name MUST not contain any spaces, or control characters.  The
   registration SHALL indicate if the feature tag applies to servers



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   only, proxies only or both server and proxies. Any proprietary
   feature SHALL have as the first part of the name a vendor tag, which
   identifies the organization.

20.1.3 Registered entries

   The following feature-tags are in this specification defined and
   hereby registered. The change control belongs to the Authors and the
   IETF MMUSIC WG.

        play.basic: The minimal implementation for playback operations
             according to section D. Applies for both servers and
             proxies.

        play.scale: Support of scale operations for media playback.
             Applies only for servers.

        play.speed: Support of the speed functionality for playback.
             Applies only for servers

20.2 RTSP Methods

20.2.1 Description

   What a method is, is described in section 11.  Extending the protocol
   with new methods allow for totally new functionality.

20.2.2 Registering New Methods with IANA

   A new method MUST be registered through an IETF standard track
   document. The reason is that new methods may radically change the
   protocols behavior and purpose.

   A specification for a new RTSP method MUST consist of the following
   items:

        o A method name which follows the BNF rules for methods.

        o A clear specification on what action and response a request
          with the method will result in. Which directions the method is
          used, C -> S or S -> C or both. How the use of headers, if
          any, modifies the behavior and effect of the method.

        o A list or table specifying which of the registered headers
          that are allowed to use with the method in request or/and
          response.

        o Describe how the method relates to network proxies.



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20.2.3 Registered Entries

   This specification, RFCXXXX, registers 10 methods: DESCRIBE,
   GET_PARAMETER, OPTIONS, PAUSE, PING, PLAY, REDIRECT, SETUP,
   SET_PARAMETER, and TEARDOWN.

20.3 RTSP Status Codes

20.3.1 Description

   A status code is the three digit numbers used to convey information
   in RTSP response messages, see  7.  The number space is limited and
   care should be taken not to fill the space.

20.3.2 Registering New Status Codes with IANA

   A new status code can only be registered by an IETF standards track
   document. A specification for a new status code MUST specify the
   following:

        o The requested number.

        o A description what the status code means and the expected
          behavior of the sender and receiver of the code.

20.3.3 Registered Entries

   RFCXXX, registers the numbered status code defined in the BNF entry
   "Status-Code" except "extension-code" in section 18.2.2.

20.4 RTSP Headers

20.4.1 Description

   By specifying new headers a method(s) can be enhanced in many
   different ways. An unknown header will be ignored by the receiving
   entity. If the new header is vital for a certain functionality, a
   feature-tag for the functionality can be created and demanded to be
   used by the counter-part with the inclusion of a Require header
   carrying the feature-tag.

20.4.2 Registering New Headers with IANA

   A public available specification is required to register a header.
   The specification SHOULD be a standards document, preferable an IETF
   RFC.

   The specification MUST contain the following information:



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        o The name of the header.

        o A BNF specification of the header syntax.

        o A list or table specifying when the header may be used,
          encompassing all methods, their request or response, the
          direction (C -> S or S -> C).

        o How the header shall be handled by proxies.

        o A description of the purpose of the header.

20.4.3 Registered entries

   All headers specified in section 14 in RFCXXXX are to be registered.

   Furthermore the following RTSP headers defined in other
   specifications are registered:

        o x-wap-profile defined in [39].

        o x-wap-profile-diff defined in [39].

        o x-wap-profile-warning defined in [39].

        o x-predecbufsize defined in [39].

        o x-initpredecbufperiod defined in [39].

        o x-initpostdecbufperiod defined in [39].

          Note: The use of "X-" is NOT RECOMMENDED but the above headers
          in the register list was defined prior to the clarification.

20.5 Transport Header registries

   The transport header contains a number of parameters which have
   possibilities for future extensions. Therefore registries for these
   must be defined.

20.5.1 Transport Protocols

   A registry for the parameter transport-protocol shall be defined with
   the following rules:

        o Registering requires public available standards specification.

        o A contact person or organization with address and email.



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        o A value definition that are following the BNF token
          definition.

        o A describing text that explains how the registered value are
          used in RTSP.

   This specification registers 1 value:

        o Use of the RTP  [18] protocol for media transport. The usage
          is explained in RFC XXXX, appendix B.1.

20.5.2 Profile

   A registry for the parameter profile shall be defined with the
   following rules:

        o Registering requires public available standards specification.

        o A contact person or organization with address and email.

        o A value definition that are following the BNF token
          definition.

        o A definition of which Transport protocol(s) that this profile
          is valid for.

        o A describing text that explains how the registered value are
          used in RTSP.

   This specification registers 1 value:

        o The "RTP profile for audio and video conferences with minimal
          control"  [3] MUST only be used when the transport
          specification's transport-protocol is "RTP".

20.5.3 Lower Transport

   A registry for the parameter lower-transport shall be defined with
   the following rules:

        o Registering requires public available standards specification.

        o A contact person or organization with address and email.

        o A value definition that are following the BNF token
          definition.

        o A text describing how the registered value are used in RTSP.



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   This specification registers 2 values:

        o Indicates the use of the "User datagram protocol"  [9] for
          media transport.

        o Indicates the use Transmission control protocol  [10] for
          media transport.

20.5.4 Transport modes

   A registry for the transport parameter mode shall be defined with the
   following rules:

        o Registering requires a IETF standard tracks document.

        o A contact person or organization with address and email.

        o A value definition that are following the BNF token
          definition.

        o A describing text that explains how the registered value are
          used in RTSP.

   This specification registers 2 values:

        o See RFC XXXX.

        o See RFC XXXX.

20.6 Cache Directive Extensions

   There exist a number of cache directives which can be sent in the
   Cache-Control header. A registry for this cache directives shall be
   defined with the following rules:

        o Registering requires a IETF standard tracks document.

        o A registration shall name a contact person.

        o Name of the directive and a definition of the value, if any.

        o Specification if it is an request or response directive.

        o A describing text that explains how the cache directive is
          used for RTSP controlled media streams.

   This specification registers the following values:




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        no-cache:

        public:

        private:

        no-transform:

        only-if-cached:

        max-stale:

        min-fresh:

        must-revalidate:

        proxy-revalidate:

        max-age:

20.7 Accept-Credentials policies

   In section  17.3.1 three policies for how to handle certificates.
   Further policies may be defined and be registered with IANA using the
   following rules:

        o Registering requires an IETF standard tracks document.

        o A registration shall name a contact person.

        o Name of the policy.

        o A describing text that explains how the policy works for
          handling the certificates.

   This specification registers the following values:

        Any

        Proxy

        User

20.8 SDP attributes

   This specification defines two SDP [2] attributes that it is
   requested that IANA register.




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   SDP Attribute ("att-field"):

        Attribute name:     range
        Long form:          Media Range Attribute
        Type of name:       att-field
        Type of attribute:  Media and session level
        Subject to charset: No
        Purpose:            RFC XXXX
        Reference:          RFC XXXX
        Values:             See ABNF definition.

        Attribute name:     control
        Long form:          RTSP control URL
        Type of name:       att-field
        Type of attribute:  Media and session level
        Subject to charset: No
        Purpose:            RFC XXXX
        Reference:          RFC XXXX
        Values:             Absolute or Relative URLs.

        Attribute name:     etag
        Long form:          Entity Tag
        Type of name:       att-field
        Type of attribute:  Media and session level
        Subject to charset: No
        Purpose:            RFC XXXX
        Reference:          RFC XXXX
        Values:             See ABNF definition





A RTSP Protocol State Machine

   The RTSP session state machine describe the behavior of the protocol
   from RTSP session initialization through RTSP session termination.

   State machine is defined on a per session basis which is uniquely
   identified by the RTSP session identifier. The session may contain
   one or more media streams depending on state. If a single media
   stream is part of the session it is in non-aggregated control. If two
   or more is part of the session it is in aggregated control.

   This state machine is one possible representation that helps explain
   how the protocol works and when different requests are allowed.  We
   find it a reasonable representation but does not mandate it, and
   other representations can be created.



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A.1 States

   The state machine contains three states, described below. For each
   state there exist a table which shows which requests and events that
   is allowed and if they will result in a state change.

        Init: Initial state no session exist.

        Ready: Session is ready to start playing.

        Play: Session is playing, i.e. sending media stream data in the
             direction S -> C.

A.2 State variables

   This representation of the state machine needs more than its state to
   work. A small number of variables are also needed and is explained
   below.

        NRM: The number of media streams part of this session.

        RP: Resume point, the point in the presentation time line at
             which a request to continue will resume from. A time format
             for the variable is not mandated.

A.3 Abbreviations

   To make the state tables more compact a number of abbreviations are
   used, which are explained below.

        IFI: IF Implemented.

        md: Media

        PP: Pause Point, the point in the presentation time line at
             which the presentation was paused.

        Prs: Presentation, the complete multimedia presentation.

        RedP: Redirect Point, the point in the presentation time line at
             which a REDIRECT was specified to occur.

        SES: Session.

A.4 State Tables

   This section contains a table for each state. The table contains all
   the requests and events that this state is allowed to act on.  The



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   events which is method names are, unless noted, requests with the
   given method in the direction client to server (C -> S). In some
   cases there exist one or more requisite. The response column tells
   what type of response actions should be performed. Possible actions
   that is requested for an event includes: response codes, e.g. 200,
   headers that MUST be included in the response, setting of state
   variables, or setting of other session related parameters. The new
   state column tells which state the state machine shall change to.

   The response to valid request meeting the requisites is normally a
   2xx (SUCCESS) unless other noted in the response column. The
   exceptions shall be given a response according to the response
   column. If the request does not meet the requisite, is erroneous or
   some other type of error occur the appropriate response code MUST be
   sent. If the response code is a 4xx the session state is unchanged. A
   response code of 3rr will result in that the session is ended and its
   state is changed to Init. A response code of 304 results in no state
   change. However there exist restrictions to when a 3xx response may
   be used. A 5xx response SHALL not result in any change of the session
   state, except if the error is not possible to recover from. A
   unrecoverable error SHALL result the ending of the session. As it in
   the general case can't be determined if it was a unrecoverable error
   or not the client will be required to test. In the case that the next
   request after a 5xx is responded with 454 (Session Not Found) the
   client shall assume that the session has been ended.

   The server will timeout the session after the period of time
   specified in the SETUP response, if no activity from the client is
   detected.  Therefore there exist a timeout event for all states
   except Init.

   In the case that NRM=1 the presentation URL is equal to the media
   URL. For NRM>1 the presentation URL MUST be other than any of the
   medias that are part of the session. This applies to all states.


   The methods in Table 13 do not have any effect on the state machine
   or the state variables. However some methods do change other session
   related parameters, for example SET_PARAMETER which will set the
   parameter(s) specified in its body.


   The initial state of the state machine, see Table 14 can only be left
   by processing a correct SETUP request. As seen in the table the two
   state variables are also set by a correct request. This table also
   shows that a correct SETUP can in some cases be redirected to another
   URL and/or server by a 3rr response.




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   Event         Prerequisite    Response
   ______________________________________________________________
   DESCRIBE      Needs REDIRECT  3rr Redirect
   DESCRIBE                      200, Session description
   OPTIONS       Session ID      200, Reset session timeout timer
   OPTIONS                       200
   SET_PARAMETER Valid parameter 200, change value of parameter
   GET_PARAMETER Valid parameter 200, return value of parameter


   Table 13: None state-machine changing events


       Action           Requisite       New State  Response

       _____________________________________________________________
       SETUP                              Ready    NRM=1, RP=0.0
       SETUP            Needs Redirect    Init     3rr Redirect
       S -> C:REDIRECT  No Session hdr    Init     Terminate all SES


   Table 14: State: Init


   Action           Requisite          New State  Response
   _____________________________________________________________________
   SETUP            New URL              Ready    NRM+=1
   SETUP            Setten up URL        Ready    Change transport param
   TEARDOWN         Prs URL,NRM>1        Init     No session hdr
   TEARDOWN         md URL,NRM=1         Init     No Session hdr, NRM=0
   TEARDOWN         md URL,NRM>1         Ready    Session hdr, NRM-=1
   PLAY             Prs URL, No range    Play     Play from RP
   PLAY             Prs URL, Range       Play     according to range
   PAUSE            Prs URL              Ready    Return PP
   S -> C:REDIRECT  Range hdr            Ready    Set RedP
   S -> C:REDIRECT  no range hdr         Init     Session is removed
   Timeout                               Init
   RedP reached                          Ready    TEARDOWN of session


   Table 15: State: Ready


   In the Ready state, see Table 15, some of the actions are depending
   on the number of media streams (NRM) in the session, i.e. aggregated
   or non-aggregated control. A setup request in the ready state can
   either add one more media stream to the session or if the media


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   stream (same URL) already is part of the session change the transport
   parameters. TEARDOWN is depending on both the request URL and the
   number of media stream within the session. If the request URL is the
   presentations URL the whole session is torn down. If a media URL is
   used in the TEARDOWN request and more than one media exist in the
   session, the session will remain and a session header MUST be
   returned in the response. If only a single media stream remains in
   the session when performing a TEARDOWN with a media URL the session
   is removed. The number of media streams remaining after tearing down
   a media stream determines the new state.


   Action           Requisite         New State  Response
   ______________________________________________________________________
   PAUSE            PrsURL,No range     Ready    Set RP to present point
   PAUSE            PrsURL,Range>now    Play     Set RP & PP to given p.
   PAUSE            PrsURL,Range<now    Ready    Set RP to Range Hdr.
   PP reached                           Ready    RP = PP
   End of media     All media           Play     No action, RP = Invalid
   End of media     >1 Media plays      Play     No action
   End of range                         Play     Set RP = End of range
   SETUP            New URL             Play     455
   SETUP            Setuped URL         Play     455
   SETUP            Setuped URL, IFI    Play     Change transport param.
   TEARDOWN         Prs URL,NRM>1       Init     No session hdr
   TEARDOWN         md URL,NRM=1        Init     No Session hdr, NRM=0
   TEARDOWN         md URL              Play     455
   S -> C:REDIRECT  Range hdr           Play     Set RedP
   S -> C:REDIRECT  no range hdr        Init     Session is removed
   RedP reached                         Play     TEARDOWN of session
   Timeout                              Init     Stop Media playout


   Table 16: State: Play


   The Play state table, see Table 16, is the largest. The table
   contains an number of request that has presentation URL as a
   prerequisite on the request URL, this is due to the exclusion of
   non-aggregated stream control in sessions with more than one media
   stream.

   To avoid inconsistencies between the client and server, automatic
   state transitions are avoided. This can be seen at for example "End
   of media" event when all media has finished playing, the session
   still remain in Play state. An explicit PAUSE request must be sent to
   change the state to Ready. It may appear that there exist two
   automatic transitions in "RedP reached" and "PP reached", however
   they are requested and acknowledge before they take place. The time


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   at which the transition will happen is known by looking at the range
   header. If the client sends request close in time to these
   transitions it must be prepared for getting error message as the
   state may or may not have changed.

B Media Transport Alternatives

   This chapter defines how certain combinations of protocols, profiles
   and lower transports are used. This includes the usage of the
   Transport header's general source and destination parameters
   "src_addr" and "dest_addr".

B.1 RTP

   This section defines the interaction of RTSP with respect to the RTP
   protocol [18]. It also defines any necessary media transport
   signalling with regards to RTP.

   The available RTP profiles and lower layer transports are described
   below along with rules on signalling the available combinations.

B.1.1 AVP

   The usage of the "RTP Profile for Audio and Video Conferences with
   Minimal Control" [3] when using RTP for media transport over
   different lower layer transport protocols are defined below in
   regards to RTSP.

   On such case is defined within this document, the use of embedded
   (interleaved) binary data as defined in section  12.  The usage of
   this method is indicated by include the "interleaved" parameter.

   When using embedded binary data the "src_addr" and "dest_addr" SHALL
   NOT be used. This addressing and multiplexing is used as defined with
   use of channel numbers and the interleaved parameter.

B.1.2 AVP/UDP

   This part describes sending of RTP [18] over lower transport layer
   UDP [9] according to the profile "RTP Profile for Audio and Video
   Conferences with Minimal Control" defined in RFC 3551 [3]. This
   profiles requires that one or two uni- or bi-directional UDP flows
   per media stream.  The first UDP flow is for RTP and the second is
   for RTCP. Embedded (interleaved) data when RTSP messages is
   transported over UDP SHOULD NOT be performed.

   The RTP/UDP and RTCP/UDP flows can be established in two ways using
   the Transport header's parameters. The way provided in RFC 2326 was



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   to use the necessary parameters from the set of "source",
   "destination", "client_port", and "server_port". This has the
   advantage of being compatible with all RTP capable RTSP servers and
   clients. However this method does not provide a possibility to
   specify non-continues port ranges for RTP and RTCP.  The other way is
   to use the parameters "src_addr", and "dest_addr". This method
   provides total flexibility in specifying address and port number for
   each transport flow. However the disadvantage is that it is not
   supported by non-updated clients, i.e. clients not supporting the
   "play.basic" feature-tag.

   When using the "source", "destination", "client_port", and
   "server_port" the packets are be addressed in the following way for
   media playback:


        o RTP/UDP packet from the server to the client SHALL be sent to   |
          the address specified in the "destination" parameter and first  |
          even port number given in client_port range. If only a RTP      |
          port is to be specified, then only that even port number SHALL  |
          be given, i.e. no range including a odd number SHALL be used.

        o The server SHOULD send its RTP/UDP packets from the address
          specified in "source" parameter and from the first even port
          number specified in "server_port" parameter.

        o If there is specified a range in "client_port" parameter that
          contains at least two port numbers, the RTCP/UDP packets from
          server to client SHALL be sent to the address specified in the
          "destination" parameter and first odd port number part of the
          range specified in the client_port parameter.

        o The Server SHOULD send its RTCP/UDP packets from the address
          specified in "source" parameter and from the first odd port
          number greater than the RTP port number specified in
          "server_port" parameter.


        o RTCP/UDP packets from the client to the server SHALL be sent    |
          to the address specified in the "source" parameter and first    |
          odd port number greater than the RTP port number given in       |
          server_port range.                                              |

        o The client SHOULD send its RTCP/UDP packets from the address    |
          specified in "destination" parameter and from the first odd     |
          port number specified in client_port" parameter.

   The usage of "src_addr" and "dest_addr" parameters to specify the



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   address and port numbers are done in the following way for media
   playback, i.e. Mode=PLAY:

        o The "src_addr" and "dest_addr" parameters MUST contain either
          1 or 2 address and port pairs.

        o Each address and port pair MUST contain both and address and a
          port number.

        o The first address and port pair given in either of the
          parameters applies to the RTP stream. The second address and
          port pair if present applies to the RTCP stream.

        o The RTP/UDP packets from the server to the client SHALL be
          sent to the address and port given by first address and port
          pair of the "dest_addr" parameter.

        o The RTCP/UDP packets from the server to the client SHALL be
          sent to the address and port given by the second address and
          port pair of the "dest_addr" parameter. If no second pair is
          given RTCP SHALL NOT be sent.


        o The RTCP/UDP packets from the client to the server SHALL be     |
          sent to the address and port given by the second address and    |
          port pair of the "src_addr" parameter. If no second pair is     |
          given RTCP SHALL NOT be sent.

        o RTP and RTCP Packets SHOULD be sent from the corresponding
          receiver port, i.e. RTCP packets from server should be sent
          from the "src_addr" parameters second address port pair.

B.1.3 AVP/TCP

   Note that this combination is not yet defined using sperate TCP
   connections. However the use of embedded (interleaved) binary data
   transported on the RTSP connection is possible as specified in
   section  12. When using this declared combination of interleaved
   binary data the RTSP messages MUST be transported over TCP.

   A possible future for this profile would be to define the use of a
   combination of the two drafts "Connection-Oriented Media Transport in
   SDP" [40] and "Framing RTP and RTCP Packets over Connection-Oriented
   Transport" [41].  However as this work is not finished, this
   functionality is unspecified.

B.1.4 Handling NPT Jumps in the RTP Media Layer




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   RTSP allows media clients to control selected, non-contiguous
   sections of media presentations, rendering those streams with an RTP
   media layer[18]. Such control allows jumps to be created in NPT
   timeline of the RTSP session. For example, jumps in NPT can be caused
   by multiple ranges in the range specifier of a PLAY request or
   through a "seek" opertaion on an RTSP session which involves a PLAY,
   PAUSE, PLAY scenario where a new NPT is set for the session. The
   media layer rendering the RTP stream should not be affected by jumps
   in NPT. Thus, both RTP sequence numbers and RTP timestamps MUST be
   continuous and monotonic across jumps of NPT.


        We cannot assume that the RTSP client can communicate with
        the RTP media agent, as the two may be independent
        processes.  If the RTP timestamp shows the same gap as the
        NPT, the media agent will assume that there is a pause in
        the presentation. If the jump in NPT is large enough, the
        RTP timestamp may roll over and the media agent may believe
        later packets to be duplicates of packets just played out.

   As an example, assume a clock frequency of 8000 Hz, a packetization
   interval of 100 ms and an initial sequence number and timestamp of
   zero.


      C->S: PLAY rtsp://xyz/fizzle RTSP/1.0
        CSeq: 4
        Session: abcdefg
        Range: npt=10-15;

      S->C: RTSP/1.0 200 OK
        CSeq: 4
        Session: abcdefg
        Range: npt=10-15
        RTP-Info: url= rtsp://xyz/fizzle/audiotrack;seq=0;
              rtptime=0



   The ensuing RTP data stream is depicted below:


      S -> C: RTP packet - seq = 0,  rtptime = 0,     NPT time = 10s
      S -> C: RTP packet - seq = 1,  rtptime = 800,   NPT time = 10.1s
       . . .
      S -> C: RTP packet - seq = 49, rtptime = 39200, NPT time = 14.9s





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   Immediately after the end of the play range, the client follows up
   with a request to PLAY from a new NPT.


   C->S: PAUSE rtsp://xyz/fizzle RTSP/1.0
         CSeq: 5
         Session: abcdefg

   S->C: RTSP/1.0 200 OK
         CSeq: 5
         Session: abcdefg
         Range: npt=15-15

   C->S: PLAY rtsp://xyz/fizzle RTSP/1.0
         CSeq: 6
         Session: abcdefg
         Range: npt=18-20;

   S->C: RTSP/1.0 200 OK
         CSeq: 6
         Session: abcdefg
         Range: npt=18-20
         RTP-Info: url= rtsp://xyz/fizzle/audiotrack;seq=50;
                   rtptime=40000



   The ensuing RTP data stream is depicted below:

      S->C: RTP packet - seq = 50, rtptime = 40000, NPT time = 18s
      S->C: RTP packet - seq = 51, rtptime = 40800, NPT time = 18.1s
       . . .
      S->C: RTP packet - seq = 69, rtptime = 55200, NPT time = 19.9s



   First we play NPT 10 through 15, then skip ahead and play NPT 18
   through 20. The first segment is presented as RTP packets with
   sequence numbers 0 through 49 and timestamp 0 through 39,200. The
   second segment consists of RTP packets with sequence number 50
   through 69, with timestamps 40,000 through 55,200. While there is a
   gap in the NPT, there is no gap in the sequence number or timestamp
   space of the RTP data stream.

B.1.5 Handling RTP Timestamps after PAUSE

   During a PAUSE / PLAY interaction in a RTSP session, the duration of
   time for which the RTP transmission was halted MUST be reflected in



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   the RTP timestamp of each RTP stream. The duration can be calculated
   for each RTP stream as the time elapsed from when the last RTP packet
   was sent before the PAUSE request was received and when the first RTP
   packet was sent after thesubsequent PLAY request was received. The
   duration includes all latency incurred and processing time required
   to complete the request.


        The RTP RFC [18] states that: The RTP timestamp for each
        unit[packet] would be related to the wallclock time at
        which the unit becomes current on the virtual presentation
        timeline.

   In order to satisfy the requirements of [18], the RTP timestamp space
   must increase continously with real time.  While this is not optimal
   for stored media, it is required for RTP and RTCP to function as
   intended. Using a continous RTP timestamp space allows the same
   timestamp model for both stored and live media and allows better
   opportunity to integrate both types of media under a single control.

   As an example, assume a clock frequency of 8000 Hz, a packetization
   interval of 100 ms and an initial sequence number and timestamp of
   zero.


   C->S: PLAY rtsp://xyz/fizzle RTSP/1.0
         CSeq: 4
         Session: abcdefg
         Range: npt=10-15;

   S->C: RTSP/1.0 200 OK
         CSeq: 4
         Session: abcdefg
         Range: npt=10-15
         RTP-Info: url= rtsp://xyz/fizzle/audiotrack;seq=0;
              rtptime=0



   The ensuing RTP data stream is depicted below:


      S -> C: RTP packet - seq = 0, rtptime = 0,    NPT time = 10s
      S -> C: RTP packet - seq = 1, rtptime = 800,  NPT time = 10.1s
      S -> C: RTP packet - seq = 2, rtptime = 1600, NPT time = 10.2s
      S -> C: RTP packet - seq = 3, rtptime = 2400, NPT time = 10.3s





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   The client then sends a PAUSE request:


   C->S: PAUSE rtsp://xyz/fizzle RTSP/1.0
         CSeq: 5
         Session: abdcdefg

   S->C: RTSP/1.0 200 OK
         CSeq: 5
         Session: abcdefg
         Range: npt=10.4-15



   20 seconds elapse and then the client sends a PLAY request. In
   addtion the server requires 15 ms to process the request:


   C->S: PLAY rtsp://xyz/fizzle RTSP/1.0
         CSeq: 6
         Session: abcdefg

   S->C: RTSP/1.0 200 OK
         CSeq: 6
         Session: abcdefg
         Range: npt=10.4-15
         RTP-Info: url= rtsp://xyz/fizzle/audiotrack;seq=4;
                 rtptime=164400



   The ensuing RTP data stream is depicted below:

      S -> C: RTP packet - seq = 4, rtptime = 164400, NPT time = 10.4s
      S -> C: RTP packet - seq = 5, rtptime = 165200, NPT time = 10.5s
      S -> C: RTP packet - seq = 6, rtptime = 166000, NPT time = 10.6s



   First, we play NPT 10 through 10.3, when a PAUSE is received by the
   server. After 20 seconds a PLAY is recieved by the server which take
   15ms to process. The duration of time for which the session was
   paused is reflected in the RTP timestamp of the RTP packets sent
   after this PLAY request.

   A client can use the RTSP range header and RTP-Info header to map NPT
   time of a presentation with the RTP timestamp.




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   Note: In RFC 2326 [1], this matter was not clearly defined and was
   misunderstood commonly. Therefore, clients SHOULD expect servers to
   break the continuity of the RTP timestamp space in various arbitrary
   manners after a PAUSE request. In these cases, it is RECOMMENDED that
   clients accept the RTP stream after the pause with appropriate
   mappings provided by the RTP-Info and Range headers.

B.1.6 RTSP / RTP Integration

   For certain datatypes, tight integration between the RTSP layer and
   the RTP layer will be necessary. This by no means precludes the above
   restrictions. Combined RTSP/RTP media clients should use the RTP-Info
   field to determine whether incoming RTP packets were sent before or
   after a seek or before or after a PAUSE.

B.1.7 Scaling with RTP

   For scaling (see Section 14.37), RTP timestamps should correspond to
   the playback timing. For example, when playing video recorded at 30
   frames/second at a scale of two and speed (Section 14.38) of one, the
   server would drop every second frame to maintain and deliver video
   packets with the normal timestamp spacing of 3,000 per frame, but NPT
   would increase by 1/15 second for each video frame.

B.1.8 Maintaining NPT synchronization with RTP timestamps

   The client can maintain a correct display of NPT by noting the RTP
   timestamp value of the first packet arriving after repositioning.
   The sequence parameter of the RTP-Info (Section 14.36) header
   provides the first sequence number of the next segment.

B.1.9 Continuous Audio

   For continuous audio, the server SHOULD set the RTP marker bit at the
   beginning of serving a new PLAY request. This allows the client to
   perform playout delay adaptation.

B.1.10 Multiple Sources in an RTP Session

   Note that more than one SSRC MAY be sent in the media stream.
   However, without further extensions RTSP can't synchronize more than
   the single one indicated in the Transport header. In these cases RTCP
   needs to be used for synchronization.

B.1.11 Usage of SSRCs and the RTCP BYE Message During a RTSP Session


   The RTCP BYE message indicates the end of use of a given SSRC. If all  |



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   sources leave an RTP session, it can in most cases be considered to    |
   have ended. Therefore, a client or server SHALL NOT send a RTCP BYE    |
   message until it has finished using a SSRC. A server SHOULD keep       |
   using a SSRC until the RTP session is terminated. Prologing the use    |
   of a SSRC allows the established synchronization context associated    |
   with that SSRC to be used to sychronize subsequent PLAY requests even  |
   if the PLAY response is late. Additionally, changing the server side   |
   SSRC will prevent the server from synchronizing the new SSRC within    |
   RTSP as it is connected to the one declared in the ssrc parameter in   |
   the Transport header.                                                  |

   An SSRC collision with the SSRC that transmits media does also has     |
   consequences, as it will force it to change its SSRC in accordance     |
   with the RTP specification  [18]. This will result in a loss of        |
   synchronization context, and require any receiver to wait for RTCP     |
   sender reports for all media requiring synchronization before being    |
   able to play out synchronized. Due to these reasons a client joining   |
   a session should take care to not select the same SSRC as the server.  |
   Any SSRC signalled in the Transport header SHOULD be avoided. Also a   |
   client detecting a collision prior to sending any RTP or RTCP          |
   messages can also select a new SSRC.                                   |


B.2 Future Additions

   It is the intention that any future protocol or profile regarding
   both for media delivery and lower transport should be easy to add to
   RTSP. This chapter provides the necessary steps that needs to be
   meet.

   The following things needs to be considered when adding a new
   protocol of profile for use with RTSP:

        o The protocol or profile needs to define a name tag
          representing it. This tag is required to be a ABNF "token" to
          be possible to use in the Transport header specification.

        o The useful combinations of protocol/profile/lower-layer needs
          to be defined and for each combination declare the necessary
          parameters to use in the Transport header.

        o For new media protocols the interaction with RTSP needs to be
          addressed. One important factor will be the media
          synchronization.

   See the IANA section ( 20) on how to register the necessary
   attributes.




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C Use of SDP for RTSP Session Descriptions

   The Session Description Protocol (SDP, RFC 2327 [2]) may be used to
   describe streams or presentations in RTSP. This description is
   typically returned in reply to a DESCRIBE request on a URL from a
   server to a client, or received via HTTP from a server to a client.

   This appendix describes how an SDP file determines the operation of
   an RTSP session. SDP as is provides no mechanism by which a client
   can distinguish, without human guidance, between several media
   streams to be rendered simultaneously and a set of alternatives
   (e.g., two audio streams spoken in different languages). However the
   SDP extension "Grouping of Media Lines in the Session Description
   Protocol (SDP)" [42] may provide such functionality depending on
   need. Also future grouping semantics may in the future be developed.

C.1 Definitions

   The terms "session-level", "media-level" and other key/attribute
   names and values used in this appendix are to be used as defined in
   SDP (RFC 2327 [2]):

C.1.1 Control URL

   The "a=control:" attribute is used to convey the control URL. This
   attribute is used both for the session and media descriptions. If
   used for individual media, it indicates the URL to be used for
   controlling that particular media stream.  If found at the session
   level, the attribute indicates the URL for aggregate control
   (presentation URL). The session level URL SHALL be different from any
   media level URL. The presence of a session level control attribute
   SHALL be interpreted as support for aggregated control. The control
   attribute SHALL be present on media level unless the presentation
   only contains a single media stream, in which case the attribute MAY
   only be present on the session level.


   control-attribute  =  "a=" "control" ":" url


   Example:

     a=control:rtsp://example.com/foo



   This attribute MAY contain either relative and absolute URLs,
   following the rules and conventions set out in RFC 2396 [13].



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   Implementations SHALL look for a base URL in the following order:

        1.   the RTSP Content-Base field;

        2.   the RTSP Content-Location field;

        3.   the RTSP request URL.

   If this attribute contains only an asterisk (*), then the URL SHALL
   be treated as if it were an empty embedded URL, and thus inherit the
   entire base URL.

   For SDP retrieved from a container file, there are certain things to
   consider. Lets say that the container file has the following URL:
   "rtsp://example.com/container.mp4". A media level relative URL needs
   to contain the file name container.mp4 in the beginning to be
   resolved correctly relative to the before given URL. An alternative
   if one does not desire to enter the container files name is to ensure
   that the base URL for the SDP document becomes:
   "rtsp://example.com/container.mp4/", i.e. an extra trailing slash.
   When using the URL resolution rules in RFC 2396 that will resolve
   correctly. However, please note that if the session level control URL
   is a *, that control URL will be equal to
   "rtsp://example.com/container.mp4/" and include the slash.

C.1.2 Media Streams

   The "m=" field is used to enumerate the streams. It is expected that
   all the specified streams will be rendered with appropriate
   synchronization. If the session is a multicast, the port number
   indicated SHOULD be used for reception. The client MAY try to
   override the destination port, through the Transport header.  The
   servers MAY allow this, the response will indicate if allowed or not.
   If the session is unicast, the port number is the ones RECOMMENDED by
   the server to the client, about which receiver ports to use; the
   client MUST still include its receiver ports in its SETUP request.
   The client MAY ignore this recommendation. If the server has no
   preference, it SHOULD set the port number value to zero.

   The "m=" lines contain information about what transport protocol,
   profile, and possibly lower-layer shall be used for the media stream.
   The combination of transport, profile and lower layer, like
   RTP/AVP/UDP needs to be defined for how to be used with RTSP.  The
   currently defined combinations are defined in section B, further
   combinations MAY be specified.

   TODO: Write something about the usage of Grouping of media line, RFC
   3388 [42].



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   Example:

     m=audio 0 RTP/AVP 31



C.1.3 Payload Type(s)

   The payload type(s) are specified in the "m=" field. In case the
   payload type is a static payload type from RFC 3551 [3], no other
   information is required. In case it is a dynamic payload type, the
   media attribute "rtpmap" is used to specify what the media is. The
   "encoding name" within the "rtpmap" attribute may be one of those
   specified in RFC 3551 (Sections 5 and 6), or an MIME type registered
   with IANA, or an experimental encoding as specified in SDP (RFC 2327
   [2]). Codec-specific parameters are not specified in this field, but
   rather in the "fmtp" attribute described below.

C.1.4 Format-Specific Parameters

   Format-specific parameters are conveyed using the "fmtp" media
   attribute. The syntax of the "fmtp" attribute is specific to the
   encoding(s) that the attribute refers to. Note that some of the
   format specific parameters may be specified outside of the fmtp
   parameters, like for example the "ptime" attribute for most audio
   encodings.

C.1.5 Range of Presentation

   The "a=range" attribute defines the total time range of the stored
   session or an individual media. Non-seekable live sessions can be
   indicated, while the length of live sessions can be deduced from the
   "t" and "r" SDP parameters.

   The attribute is both a session and a media level attribute. For
   presentations that contains media streams of the same durations, the
   range attribute SHOULD only be used at session-level. In case of
   different length the range attribute MUST be given at media level for
   all media, and SHOULD NOT be given at session level. If the attribute
   is present at both media level and session level the media level
   values SHALL be used.

   The unit is specified first, followed by the value range. The units
   and their values are as defined in Section 3.4, 3.5 and 3.6 and MAY
   be extended with further formats. Any open ended range (start-), i.e.
   without stop range, is of unspecified duration and SHALL be
   considered as non-seekable content unless this property is
   overridden.



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   This attribute is defined in ABNF [5] as:

   a-range-def = "a" "=" "range" ":" ranges-specifier CRLF


   Examples:

     a=range:npt=0-34.4368
     a=range:clock=19971113T2115-19971113T2203
     Non seekable stream of unknown duration:
     a=range:npt=0-



C.1.6 Time of Availability

   The "t=" field MUST contain suitable values for the start and stop
   times for both aggregate and non-aggregate stream control.  The
   server SHOULD indicate a stop time value for which it guarantees the
   description to be valid, and a start time that is equal to or before
   the time at which the DESCRIBE request was received. It MAY also
   indicate start and stop times of 0, meaning that the session is
   always available.

   For sessions that are of live type, i.e. specific start time, unknown
   stop time, likely unseekable, the "t=" and "r=" field SHOULD be used
   to indicate the start time of the event. The stop time SHOULD be
   given so that the live event will with high probability have ended at
   that time, while still not be unnecessary long into the future.

C.1.7 Connection Information

   In SDP, the "c=" field contains the destination address for the media
   stream. For a media destination address that is a IPv6 one, the SDP
   extension defined in [22] needs to be used. For on-demand unicast
   streams and some multicast streams, the destination address MAY be
   specified by the client via the SETUP request, thus overriding any
   specified address. To identify streams without a fixed destination
   address, where the client must specify a destination address, the
   "c=" field SHOULD be set to a null value. For addresses of type
   "IP4", this value SHALL be "0.0.0.0", and for type "IP6", this value
   SHALL be "0:0:0:0:0:0:0:0", i.e. the unspecified address according to
   RFC 3513 [23].

C.1.8 Entity Tag

   The optional "a=etag" attribute identifies a version of the session
   description. It is opaque to the client. SETUP requests may include



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   this identifier in the If-Match field (see section 14.24) to only
   allow session establishment if this attribute value still corresponds
   to that of the current description.  The attribute value is opaque
   and may contain any character allowed within SDP attribute values.


   a-etag-def   =  "a" "=" "etag" ":" etag-string CRLF
   etag-string  =  1*(%x01-09/%x0B-0C/%x0E-FF)


   Example:

     a=etag:158bb3e7c7fd62ce67f12b533f06b83a




        One could argue that the "o=" field provides identical
        functionality. However, it does so in a manner that would
        put constraints on servers that need to support multiple
        session description types other than SDP for the same piece
        of media content.

C.2 Aggregate Control Not Available

   If a presentation does not support aggregate control no session level
   "a=control:" attribute is specified. For a SDP with multiple media
   sections specified, each section will have its own control URL
   specified via the "a=control:" attribute.

   Example:

   v=0
   o=- 2890844256 2890842807 IN IP4 204.34.34.32
   s=I came from a web page
   e=adm@example.com
   c=IN IP4 0.0.0.0
   t=0 0
   m=video 8002 RTP/AVP 31
   a=control:rtsp://audio.com/movie.aud
   m=audio 8004 RTP/AVP 3
   a=control:rtsp://video.com/movie.vid



   Note that the position of the control URL in the description implies
   that the client establishes separate RTSP control sessions to the
   servers audio.com and video.com



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   It is recommended that an SDP file contains the complete media
   initialization information even if it is delivered to the media
   client through non-RTSP means. This is necessary as there is no
   mechanism to indicate that the client should request more detailed
   media stream information via DESCRIBE.

C.3 Aggregate Control Available

   In this scenario, the server has multiple streams that can be
   controlled as a whole. In this case, there are both a media-level
   "a=control:" attributes, which are used to specify the stream URLs,
   and a session-level "a=control:" attribute which is used as the
   request URL for aggregate control. If the media-level URL is
   relative, it is resolved to absolute URLs according to Section C.1.1
   above.

   Example:


   C->M: DESCRIBE rtsp://example.com/movie RTSP/1.0
         CSeq: 1

   M->C: RTSP/1.0 200 OK
         CSeq: 1
         Date: 23 Jan 1997 15:35:06 GMT
         Content-Type: application/sdp
         Content-Base: rtsp://example.com/movie/
         Content-Length: 164

         v=0
         o=- 2890844256 2890842807 IN IP4 204.34.34.32
         s=I contain
         i=<more info>
         e=adm@example.com
         c=IN IP4 0.0.0.0
         t=0 0
         a=control:*
         m=video 8002 RTP/AVP 31
         a=control:trackID=1
         m=audio 8004 RTP/AVP 3
         a=control:trackID=2



   In this example, the client is required to establish a single RTSP
   session to the server, and uses the URLs
   rtsp://example.com/movie/trackID=1 and
   rtsp://example.com/movie/trackID=2 to set up the video and audio



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   streams, respectively. The URL rtsp://example.com/movie/ , which is
   resolved from the "*", controls the whole presentation (movie).

   A client is not required to issues SETUP requests for all streams
   within an aggregate object. Servers should allow the client to ask
   for only a subset of the streams.

C.4 RTSP external SDP delivery

   There are some considerations that needs to be made when the session
   description is delivered to client outside of RTSP, for example in
   HTTP or email.

   First of all the SDP needs to contain absolute URLs, relative will in
   most cases not work as the delivery will not correctly forward the
   base URL. And as SDP might be temporarily stored on file system
   before being loaded into a RTSP capable client, thus if possible to
   transport the base URL it still would need to be merged into the
   file.

   The writing of the SDP session availability information, i.e. "t="
   and "r=", needs to be carefully considered. When the SDP is fetched
   by the DESCRIBE method it is with very high probability that the it
   is valid. However the same are much less certain for SDPs distributed
   using other methods. Therefore the publisher of the SDP should take
   care to follow the recommendations about availability in the SDP
   specification [2].

D Minimal RTSP implementation

D.1 Client

   A client implementation MUST be able to do the following :

        o Generate the following requests: SETUP, TEARDOWN, PLAY.

        o Include the following headers in requests: CSeq, Connection,
          Session, Transport.

        o Parse and understand the following headers in responses:
          CSeq, Connection, Session, Transport, Content-Language,
          Content-Encoding, Content-Length, Content-Type.

        o Understand the class of each error code received and notify
          the end-user, if one is present, of error codes in classes 4xx
          and 5xx. The notification requirement may be relaxed if the
          end-user explicitly does not want it for one or all status
          codes.



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        o Expect and respond to asynchronous requests from the server,
          such as REDIRECT. This does not necessarily mean that it
          should implement the REDIRECT method, merely that it MUST
          respond positively or negatively to any request received from
          the server.

   Though not required, the following are RECOMMENDED.

        o Implement RTP/AVP/UDP as a valid transport.

        o Inclusion of the User-Agent header.

        o Understand SDP session descriptions as defined in Appendix C

        o Accept media initialization formats (such as SDP) from
          standard input, command line, or other means appropriate to
          the operating environment to act as a "helper application" for
          other applications (such as web browsers).


        There may be RTSP applications different from those
        initially envisioned by the contributors to the RTSP
        specification for which the requirements above do not make
        sense. Therefore, the recommendations above serve only as
        guidelines instead of strict requirements.

D.1.1 Basic Playback

   To support on-demand playback of media streams, the client MUST
   additionally be able to do the following:

        o generate the PAUSE request;

        o implement the REDIRECT method, and the Location header.

D.1.2 Authentication-enabled

   In order to access media presentations from RTSP servers that require
   authentication, the client MUST additionally be able to do the
   following:

        o recognize the 401 (Unauthorized) status code;

        o parse and include the WWW-Authenticate header;

        o implement Basic Authentication and Digest Authentication.

D.2 Server



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   A minimal server implementation MUST be able to do the following:

        o Implement the following methods: SETUP, TEARDOWN, OPTIONS and
          PLAY.

        o Include the following headers in responses:  Connection,
          Content-Length, Content-Type, Content-Language, Content-
          Encoding, Timestamp, Transport, Public, and Via, and
          Unsupported.  RTP-compliant implementations MUST also
          implement the RTP-Info field.

        o Parse and respond appropriately to the following headers in
          requests: Connection, Proxy-Require, Session, Transport, and
          Require.

   Though not required, the following are highly recommended at the time
   of publication for practical interoperability with initial
   implementations and/or to be a "good citizen".

        o Implement RTP/AVP/UDP as a valid transport.

        o Inclusion of the Server, Cache-Control Date, and Expires
          headers.

        o Implement the DESCRIBE method.

        o Generate SDP session descriptions as defined in Appendix C


        There may be RTSP applications different from those
        initially envisioned by the contributors to the RTSP
        specification for which the requirements above do not make
        sense. Therefore, the recommendations above serve only as
        guidelines instead of strict requirements.

D.2.1 Basic Playback

   To support on-demand playback of media streams, the server MUST
   additionally be able to do the following:

        o Recognize the Range header, and return an error if seeking is
          not supported.

        o Implement the PAUSE method.

   In addition, in order to support commonly-accepted user interface
   features, the following are highly recommended for on-demand media
   servers:



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        o Include and parse the Range header, with NPT units.
          Implementation of SMPTE units is recommended.

        o Include the length of the media presentation in the media
          initialization information.

        o Include mappings from data-specific timestamps to NPT. When
          RTP is used, the rtptime portion of the RTP-Info field may be
          used to map RTP timestamps to NPT.


        Client implementations may use the presence of length
        information to determine if the clip is seekable, and
        visably disable seeking features for clips for which the
        length information is unavailable. A common use of the
        presentation length is to implement a "slider bar" which
        serves as both a progress indicator and a timeline
        positioning tool.

   Mappings from RTP timestamps to NPT are necessary to ensure correct
   positioning of the slider bar.

D.2.2 Authentication-enabled

   In order to correctly handle client authentication, the server MUST
   additionally be able to do the following:

        o Generate the 401 (Unauthorized) status code when
          authentication is required for the resource.

        o Parse and include the WWW-Authenticate header

        o Implement Basic Authentication and Digest Authentication

E Open Issues

   This section contains a list of open issues that still needs to be
   resolved. However also any open issues in the bug tracker at
   http://rtspspec.sourceforge.net should also be considered.

        1.   The proxy indications in the two header tables in chapter
             14 needs review.

        2.   Should the Allow header be possible to use optional in
             request or responses besides the now specified 405 error
             code?

        3.   The minimal implementation must be checked to see if it



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             complies with the specification. All must and should shall
             be included in the minimal. Feature-tags for these needs to
             be defined. Further feature-tags needs to be discussed.

        4.   The list specifying which status codes are allowed on which
             request methods seem to be in error and need review.

        5.   Can fragment be included in a request URL? Or should it as
             for HTTP only be handled on the User-Agent side?

        6.   ABNF Syntax needs to be run through verifier.

        7.   There is need for clearer rule in regards to Transport
             parameters changes in mid session. It needs to be
             determined if there should be any clarification on how and
             which Transport header parameters that may be changed.

        8.   Normative suggestion is needed for doing RTSP session keep
             alives. Currently there are too many options being
             suggested by RTSP such as OPTIONS with Session ID, PING,
             SET_PARAMETER. This leads to interoperability problems,
             maintenance issues and additional development for
             implementers for little gain.

F Changes

   Compared to RFC 2326, the following issues has been addressed:

        o The Transport header has been changed in the following way:

          - The ABNF has been changed to define that extensions are
            possible, and that unknown extension parameters shall be
            ignored.

          - To prevent backwards compatibility issues, any extension or
            new parameter requires the usage of a feature tag combined
            with the Require header.

          - Syntax unclarities with the Mode parameter has been
            resolved.

          - Syntax error with ";" for multicast and unicast has been
            resolved.

          - Two new addressing parameters has been defined, src_addr and
            dest_addr. These allow one to specify more than one complete
            address and port tuple if needed.




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          - Support for IPv6 explicit addresses in all address fields
            has been included.

          - To handle URI definitions that contain ";" or "," a quoted
            URL format has been introduced.

          - Defined IANA registries for the transport headers
            parameters, transport-protocol, profile, lower-transport,
            and mode.

          - The transport headers interleaved parameter's text was made
            more strict and use formal requirements levels. However no
            change on how it is used was made.

          - It has been clarified that the client can't request of the
            server to use a certain RTP SSRC, using a request with the
            transport parameter SSRC.

          - Syntax defintion for SSRC has been clarified to require 8*8
            HEX. It has also been extend to allow multiple values for
            clients supporting this version.

          - Updated the text on the transport headers "destination" and
            "dest_addr" parameters regarding what security precautions
            the server shall perform.

          - The embedded (interleaved) binary data and its transport
            parameter was clarified to being symmetric and that it is
            the server that sets the channel numbers.

        o The Range formats has been changed in the following way:

          - The NPT format has been given a initial NPT identifier that
            should be used, if missing NPT is assumed.

          - All formats now support initial open ended formats of type
            "npt=-10".

        o RTSP message handling has been changed in the following way:

          - It has been clarified that a 4xx message due to missing CSeq
            header shall be returned without a CSeq header.

          - Rules for how to handle timing out RTSP messages has been
            added.

        o The HTTP references has been updated to RFC 2616 and RFC 2617.
          This has required that public, and content-base header are now



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          defined in the RTSP specification. Known effects on RTSP due
          to HTTP clarifications:

          - Content-Encoding header can include encoding of type
            "identity".

        o The state machine chapter has completely been rewritten. It
          includes now more details and are also more clear about the
          model used.

        o A IANA section has been included with contains a number of
          registries and their rules. This will allow us to use IANA to
          keep track of all RTSP extensions.

        o Than transport of RTSP messages has seen the following
          changes:

          - The use of UDP has been deprecated due to missing interest
            and to broken specification.

          - The rules for how TCP connections shall be handled has been
            clarified. Now it is made clear that servers should not
            close the TCP connection unless they have been unused for
            significant time.

          - Strong recommendations why server and clients should use
            persistent connections has also been added.

          - There is now a requirement to handle non-persistent
            connections as this provides great fault tolerance.

          - Added wording on the usage of Connection:Close for RTSP.

          - specified usage of TLS for RTSP messages, including a scheme
            to approve a proxies TLS connection to the next hop.

        o The following header related changes have been made:

          - Accept-Ranges response header is added. This header
            clarifies which range formats that can be used for a
            resource.

          - Clarified that Range header allows multiple ranges to allow
            for creating editing list.

          - Fixed the missing definitions for the Cache-Control header.
            Also added to the syntax definition the missing delta-
            seconds for max-stale and min-fresh parameters.



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          - Put requirement on CSeq header that the value is increased
            by one for each new RTSP request. A Recommendation to start
            at 1 has also been added.

          - Added requirement that the Date header must be used for all
            messages with entity. Also the Server should always include
            it.

          - Removed possibility of using Range header with Scale header
            to indicate when it shall be activated, since it can't work
            as defined. Also added rule that lack of Scale header in
            response indicates lack of support for the header. Feature-
            tags for scaled playback has been defined.

          - The Speed header must now be responded to indicate support
            and the actual speed going to be used. A feature-tag is
            defined. Notes on congestion control was also added.

          - The Supported header was borrowed from SIP to help with the
            feature negotiation in RTSP.

          - Clarified that the Timestamp header can be used to resolve
            retransmission ambiguities.

          - The Session header text has been expanded with a explanation
            on keep alive and which methods to use.

          - It has been clarified how the Range header formats shall be
            used to indicate pause points.

          - Clarified that RTP-Info URLs that are relative, uses the
            request URL as base URL. Also clarified that the URL that
            must be used is the SETUP.

          - Added text that requires the Range to always be present in
            PLAY responses. Clarified what should be sent in case of
            live streams.

          - The quoted URL format may also be used with the RTP-Info
            header. Backwards compatibility issues exist with such
            usage, thus it can only be used for implementations
            following this specification.

          - The headers table has been updated using a structured
            borrowed from SIP. This table carries much more information
            and should provide a good overview of the available headers.

          - It has been is clarified that any message with a message



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            body is required to have a Content-Length header. This was
            the case in RFC 2326 but could be misinterpreted.

          - To resolve functionality around ETag. The ETag and If-None-
            Match header has been added from HTTP with necessary
            clarification in regards to RTSP operation.

          - Imported the Public header from HTTP RFC 2068 [20] since it
            has been removed from HTTP due to lack of use. Public is
            used quite frequently in RTSP.

          - Clarified rules for populating the Public header so that it
            is an intersection of the capabilities of all the RTSP
            agents in a chain.

        o The minimal implementation specification has been changed:

          - Required Timestamp, Via, and Unsupported headers for a
            minimal server implementation.

          - Recommended that Cache-Control, Expires and Date headers be
            supported by server implementations.

        o The Protocol Syntax has been changed in the following way:

          - All BNF definitions are updated according to the rules
            defined in RFC 2234 [5] and has been gathered in a separate
            chapter  18.

          - The BNF for the User-Agent and Server headers has been
            corrected so now only the description is in the HTTP
            specification.

          - The definition in the introduction of the RTSP session has
            been changed.

          - The protocol has been made fully IPv6 capable. Certain of
            the functionality, like using explicit IPv6 addresses in
            fields requires that the protocol support this updated
            specification.

          - Added a fragment part to the RTSP URL. This seem to be
            indicated by the note below the definition however it was
            not part of the BNF.

        o The Status codes has been changed in the following way:

          - The use of status code 303 "See Other" has been deprecated



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            as it does not make sense to use in RTSP.

          - When sending response 451 and 458 the response body should
            contain the offending parameters.

          - Clarification on when a 3rr redirect status code can be
            received has been added. This includes receiving 3rr as a
            result of request within a established session. This
            provides clarification to a previous unspecified behavior.

          - Removed the 250 (Low On Storage Space) status code as it
            only is relevant to recording which is deprecated.

        o The following functionality has been deprecated from the
          protocol:

          - The use of Queued Play.

          - The use of PLAY method for keep-alive in play state.

          - The RECORD and ANNOUNCE methods and all related
            functionality. Some of the syntax has been removed.

          - The possibility to use timed execution of methods with the
            time parameter in the Range header.

          - The description on how rtspu works is not part of the core
            specification and will require external description. Only
            that it exist is defined here.

        o Text specifying the special behavior of PLAY for live content.

        o The following changes has been made in relation to methods:

          - The OPTIONS method has been clarified with regards to the
            use of the Public and Allow headers.

          - The RECORD and ANNOUNCE methods are removed as they are
            lacking implementation and not considered necessary in the
            core specification. Any work on these methods should be done
            as a extension document to RTSP.

          - Added text clarifying the usage of SET_PARAMETER for keep-
            alive and usage without any body.

          - Added a backwards compatibility resolution for how to handle
            the new state machine without automatic state transition,
            for example for returning to ready when finished playing.



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        o Wrote a new chapter about how to setup different media
          transport alternatives and their profiles, and lower layer
          protocols. This resulted that the appendix on RTP interaction
          was moved there instead in the part describing RTP. The
          chapter also includes guidelines what to think of when writing
          usage guidelines for new protocols and profiles.

        o Added a new chapter describing the available mechanisms to
          determine if functionality is supported, called "Capability
          Handling". Renamed option-tags to feature-tags.

        o Added a contributors chapter with people who has contribute
          actual text to the specification.

        o Added a chapter Use Cases that describes the major use cases
          for RTSP.

        o Clarified the usage of a=range and how to indicate live
          content that are not seekable with this header.

   Note that this list does not reflect minor changes in wording or
   correction of typographical errors.

   A word-by-word diff from RFC 2326 can be found at http://rtsp.org/

G Author Addresses

   Henning Schulzrinne
   Dept. of Computer Science
   Columbia University
   1214 Amsterdam Avenue
   New York, NY 10027
   USA
   electronic mail: schulzrinne@cs.columbia.edu

   Anup Rao
   Cisco
   USA
   electronic mail: anrao@cisco.com

   Robert Lanphier
   RealNetworks
   P.O. Box 91123
   Seattle, WA 98111-9223
   USA
   electronic mail: robla@real.com

   Magnus Westerlund



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   Ericsson AB, EAB/TVA/A
   Torshamsgatan 23
   SE-164 80 STOCKHOLM
   SWEDEN
   electronic mail: magnus.westerlund@ericsson.com

   Aravind Narasimhan
   Princeton, NJ
   USA
   electronic mail: aravind.narasimhan@gmail.com

H Contributors

   The following people has made written contribution included in the
   specification:

        o Tom Marshall has contributed with text about the usage of 3rr
          status codes.

        o Thomas Zheng has contributed with text regarding the usage of
          the Range in PLAY responses.

        o Sean Sheedy has contributed the text regarding the timing out
          of RTSP messages.

        o Fredrik Lindholm has contributed with text for the RTSP
          security framework.

I Acknowledgements

   This draft is based on the functionality of the original RTSP draft
   submitted in October 1996. It also borrows format and descriptions
   from HTTP/1.1.

   This document has benefited greatly from the comments of all those
   participating in the MMUSIC-WG. In addition to those already
   mentioned, the following individuals have contributed to this
   specification:

   Rahul Agarwal, Jeff Ayars, Milko Boic, Torsten Braun, Brent Browning,
   Bruce Butterfield, Steve Casner, Francisco Cortes, Kelly Djahandari,
   Martin Dunsmuir, Eric Fleischman, Jay Geagan, Andy Grignon, V.
   Guruprasad, Peter Haight, Mark Handley, Brad Hefta-Gaub, Volker Hilt,
   John K. Ho, Go Hori, Philipp Hoschka, Anne Jones, Anders Klemets,
   Ruth Lang, Stephanie Leif, Jonathan Lennox, Eduardo F. Llach, Thomas
   Marshall, Rob McCool, David Oran, Joerg Ott, Maria Papadopouli, Sujal
   Patel, Ema Patki, Alagu Periyannan, Colin Perkins, Igor Plotnikov,
   Jonathan Sergent, Pinaki Shah, David Singer, Lior Sion, Jeff Smith,



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   Alexander Sokolsky, Dale Stammen, John Francis Stracke, Maureen
   Chesire, David Walker, and Geetha Srikantan.

J Normative References

   [1] H. Schulzrinne, A. Rao, and R. Lanphier, "Real time streaming
   protocol (RTSP)," RFC 2326, Internet Engineering Task Force, Apr.
   1998.

   [2] M. Handley and V. Jacobson, "SDP: session description protocol,"
   RFC 2327, Internet Engineering Task Force, Apr. 1998.

   [3] H. Schulzrinne and S. Casner, "RTP profile for audio and video
   conferences with minimal control," RFC 3551, Internet Engineering
   Task Force, July 2003.

   [4] R. Fielding, J. Gettys, J. C. Mogul, H. Frystyk, L. Masinter, P.
   J. Leach, and T. Berners-Lee, "Hypertext transfer protocol --
   HTTP/1.1," RFC 2616, Internet Engineering Task Force, June 1999.

   [5] "Augmented BNF for syntax specifications: ABNF," RFC 2234,
   Internet Engineering Task Force, Nov. 1997.

   [6] S. Bradner, "Key words for use in RFCs to indicate requirement
   levels," RFC 2119, Internet Engineering Task Force, Mar. 1997.

   [7] T. Dierks and C. Allen, "The TLS protocol version 1.0," RFC 2246,
   Internet Engineering Task Force, Jan. 1999.

   [8] J. Franks, P. Hallam-Baker, J. Hostetler, S. Lawrence, P. J.
   Leach, A. Luotonen, and L. Stewart, "HTTP authentication: Basic and
   digest access authentication," RFC 2617, Internet Engineering Task
   Force, June 1999.

   [9] J. B. Postel, "User datagram protocol," RFC 768, Internet
   Engineering Task Force, Aug. 1980.

   [10] J. B. Postel, "Transmission control protocol," RFC 793, Internet
   Engineering Task Force, Sept. 1981.

   [11] R. Elz, "A compact representation of IPv6 addresses," RFC 1924,
   Internet Engineering Task Force, Apr. 1996.

   [12] R. Hinden, B. E. Carpenter, and L. Masinter, "Format for literal
   IPv6 addresses in URL's," RFC 2732, Internet Engineering Task Force,
   Dec. 1999.

   [13] T. Berners-Lee, R. Fielding, and L. Masinter, "Uniform resource



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   identifiers (URI): generic syntax," RFC 2396, Internet Engineering
   Task Force, Aug.  1998.

   [14] F. Yergeau, "UTF-8, a transformation format of ISO 10646," RFC
   2279, Internet Engineering Task Force, Jan. 1998.

   [15] "Requirements for Internet hosts - application and support," RFC
   1123, Internet Engineering Task Force, Oct. 1989.

   [16] NIST, "Fips pub 180-1:secure hash standard," tech. rep.,
   National Institute of Standards and Technology, Apr. 1995.

   [17] R. Housley, W. Polk, W. Ford, and D. Solo, "Internet X.509
   public key infrastructure certificate and certificate revocation list
   (CRL) profile," RFC 3280, Internet Engineering Task Force, Apr. 2002.

   [18] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP:
   a transport protocol for real-time applications," RFC 3550, Internet
   Engineering Task Force, July 2003.

   [19] E. Rescorla, "HTTP over TLS," RFC 2818, Internet Engineering
   Task Force, May 2000.

   [20] R. Fielding, J. Gettys, J. C. Mogul, H. Frystyk, and T.
   Berners-Lee, "Hypertext transfer protocol -- HTTP/1.1," RFC 2068,
   Internet Engineering Task Force, Jan. 1997.

   [21] T. Narten and H. Alvestrand, "Guidelines for writing an IANA
   considerations section in RFCs," RFC 2434, Internet Engineering Task
   Force, Oct. 1998.

   [22] S. Olson, G. Camarillo, and A. B. Roach, "Support for IPv6 in
   session description protocol (SDP)," RFC 3266, Internet Engineering
   Task Force, June 2002.

   [23] R. Hinden and S. E. Deering, "Internet protocol version 6 (ipv6)
   addressing architecture," RFC 3513, Internet Engineering Task Force,
   Apr. 2003.

K Informative References

   [24] T. Z. M. Westerlund, "How to make real-time streaming protocol
   (rtsp) traverse network address translators (nat) and interact with
   firewalls.," internet draft, Internet Engineering Task Force, Feb.
   2004.  Work in progress.

   [25] A. Narasimhan, "Mute and unmute extension to rtsp," internet
   draft, Internet Engineering Task Force, Feb. 2002.  Work in progress.



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   [26] P. Gentric, "Rtsp stream switching," internet draft, Internet
   Engineering Task Force, Jan. 2004.  Work in progress.

   [27] A. L. G. Srikantan, J. Murata, "Streaming relays," internet
   draft, Internet Engineering Task Force, Dec. 2003.  Work in progress.

   [28] F. Yergeau, G. Nicol, G. C. Adams, and M. Duerst,
   "Internationalization of the hypertext markup language," RFC 2070,
   Internet Engineering Task Force, Jan.  1997.

   [29] ISO/IEC, "Information technology -- generic coding of moving
   pictures and associated audio informaiton -- part 6: extension for
   digital storage media and control," Draft International Standard ISO
   13818-6, International Organization for Standardization ISO/IEC
   JTC1/SC29/WG11, Geneva, Switzerland, Nov. 1995.

   [30] C. Partridge and R. Hinden, "Version 2 of the reliable data
   protocol (RDP)," RFC 1151, Internet Engineering Task Force, Apr.
   1990.

   [31] H. Schulzrinne, "A comprehensive multimedia control architecture
   for the Internet," in Proc. International Workshop on Network and
   Operating System Support for Digital Audio and Video (NOSSDAV), (St.
   Louis, Missouri), May 1997.

   [32] International Telecommunication Union, "Visual telephone systems
   and equipment for local area networks which provide a non-guaranteed
   quality of service," Recommendation H.323, Telecommunication
   Standardization Sector of ITU, Geneva, Switzerland, May 1996.

   [33] P. McMahon, "GSS-API authentication method for SOCKS version 5,"
   RFC 1961, Internet Engineering Task Force, June 1996.

   [34] J. Miller, P. Resnick, and D. Singer, "Rating services and
   rating systems (and their machine readable descriptions),"
   Recommendation REC-PICS-services-961031, W3C (World Wide Web
   Consortium), Boston, Massachusetts, Oct. 1996.

   [35] J. Miller, T. Krauskopf, P. Resnick, and W. Treese, "PICS label
   distribution label syntax and communication protocols,"
   Recommendation REC-PICS-labels-961031, W3C (World Wide Web
   Consortium), Boston, Massachusetts, Oct. 1996.

   [36] R. Braden, "T/TCP -- TCP extensions for transactions functional
   specification," RFC 1644, Internet Engineering Task Force, July 1994.

   [37] W. R. Stevens, TCP/IP illustrated: the implementation, vol. 2.
   Reading, Massachusetts: Addison-Wesley, 1994.



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   [38] S. Josefsson and I. W. Ed., "The base16, base32, and base64 data
   encodings," RFC 3548, Internet Engineering Task Force, July 2003.

   [39] Third Generation Partnership Project (3GPP), "Transparent end-
   to-end packet-switched streaming service (pss); protocols and
   codecs," Technical Specification 26.234, Third Generation Partnership
   Project (3GPP), Dec. 2002.

   [40] D. Yon, "Connection-oriented media transport in sdp," internet
   draft, Internet Engineering Task Force, Mar. 2003.  Work in progress.

   [41] J. Lazzaro, "Framing rtp and rtcp packets over connection-
   oriented transport," internet draft, Internet Engineering Task Force,
   Oct. 2003.  Work in progress.

   [42] G. Camarillo, G. Eriksson, J. Holler, and H. Schulzrinne,
   "Grouping of media lines in the session description protocol (SDP),"
   RFC 3388, Internet Engineering Task Force, Dec. 2002.


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